Commit Graph

25180 Commits

Author SHA1 Message Date
1c7f5f63d1 Add SetKeyFrameRequestCallback to MediaTransportInterface
And implemented in LoopbackMediaTransport.

Bug: webrtc:9719
Change-Id: I68b16c2b6ed5583ffe9a5266e3d4cb1d94afbb97
Reviewed-on: https://webrtc-review.googlesource.com/c/113523
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25948}
2018-12-10 14:01:31 +00:00
f04feee41e Remove redundant return-statement in VCMGenericEncoder::RequestFrame
Bug: None
Change-Id: I0da8747729ec309a37146397d6bc1f32bf22c329
Reviewed-on: https://webrtc-review.googlesource.com/c/113660
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25947}
2018-12-10 13:54:39 +00:00
a1eb9c7e9b Convert NetEq tests to not use RegisterExternalDecoder.
This change converts all tests but CodecInternalCng and
DecodingErrorDuringInternalCng, which depend on the obsolete Decode
method.

Bug: webrtc:10080
Change-Id: I34b068b3aa7139ed24bd63b417a5adcfc1de7922
Reviewed-on: https://webrtc-review.googlesource.com/c/113506
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25946}
2018-12-10 13:01:21 +00:00
8b9b5f98db Activate/deactivate VP9 spatial layers.
* Stop encoding spatial layers S(n >= N) if application deactivates
spatial layer N by setting RTCRtpEncodingParameters.active = false.

* Move calculation of padding bitrate to SvcRateAllocator class.

* Pad up to minimum required bitrate of base layer if ALR probing is
enabled.

Bug: webrtc:9350
Change-Id: I398284c943d43348def535c83263fc234c9767fa
Reviewed-on: https://webrtc-review.googlesource.com/c/113240
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25945}
2018-12-10 12:55:51 +00:00
b47ccc38e7 Add chroma siting to ColorSpace
Bug: webrtc:8651
Change-Id: I82263e8b6cdcc3ebf699f5e3ebbde04e46982efb
Reviewed-on: https://webrtc-review.googlesource.com/c/113424
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25944}
2018-12-10 11:19:35 +00:00
1ec2a16121 Revert "Replace VideoEncoderFactory::QueryVideoEncoder with VideoEncoder::GetEncoderInfo"
This reverts commit cdc5eb0de179dcc866ef770ea303879c64466879.

Reason for revert: Causes wrong CPU adaptation to be used for some HW codecs since GetEncoderInfo() is polled before InitEncode().

Original change's description:
> Replace VideoEncoderFactory::QueryVideoEncoder with VideoEncoder::GetEncoderInfo
> 
> Make implementation of VideoEncoderFactory::QueryVideoEncoder optional
> until it is removed downstream and remove all implementations of it.
> 
> Bug: webrtc:10065
> Change-Id: Ibb1f9612234e536651ce53f05ee048a5d172a41f
> Reviewed-on: https://webrtc-review.googlesource.com/c/113065
> Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25924}

TBR=brandtr@webrtc.org,sakal@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org,srte@webrtc.org,perkj@webrtc.org,mirtad@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:10065
Change-Id: Idaa452e1d8c1c58cdb4ec69b88fce9042589cc3c
Reviewed-on: https://webrtc-review.googlesource.com/c/113800
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25943}
2018-12-10 10:36:00 +00:00
6a8727bd2a Update connection states to match spec changes.
These changes simplify the code, and also fix the issue where the peerconnectionstate would sometimes return to "new" during connection setup.

Bug: webrtc:9308
Change-Id: I895cd2f94a2b9688c821cca64d1a077317b99d44
Reviewed-on: https://webrtc-review.googlesource.com/c/111964
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25942}
2018-12-10 10:01:24 +00:00
10a58016ee Output plots for new DTLS events.
Bug: webrtc:10101
Change-Id: Ida8084549bc386b91fec468026c3f4a261a4ef50
Reviewed-on: https://webrtc-review.googlesource.com/c/113462
Commit-Queue: Zach Stein <zstein@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25941}
2018-12-07 21:45:10 +00:00
a59db7481c Remove unnecessary includes of common_types.h
Bug: webrtc:7626
Change-Id: I2d9275e5dc8eea6419d3c80cd68c4a01deafa9b7
Reviewed-on: https://webrtc-review.googlesource.com/c/113524
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25940}
2018-12-07 21:21:13 +00:00
ff71a49b30 Reduce transaction ids independent of host byte order.
Bug: webrtc:9972
Change-Id: I91df2f2c4854bec6d581c3beb9f57235a1ce47b1
Reviewed-on: https://webrtc-review.googlesource.com/c/112926
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Zach Stein <zstein@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25939}
2018-12-07 20:30:03 +00:00
168456c128 Enable authentication of the header as an optional WebRTC trial.
TBR=asapersson@webrtc.org

Bug: webrtc:10103
Change-Id: I3dce3cd06afab62cc30761395299dbb1c02ae444
Reviewed-on: https://webrtc-review.googlesource.com/c/113464
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25938}
2018-12-07 20:23:43 +00:00
a956d498a7 Only create ALR detector in PacedSender if deprecated functions are called.
Bug: webrtc:10108
Change-Id: Ic41693c4017b47093fc373547d59b7723493c70d
Reviewed-on: https://webrtc-review.googlesource.com/c/113527
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25937}
2018-12-07 17:50:36 +00:00
1d61c430d9 desktopCapture: copy whole screen region when screen is zoomed on OSX
When screen is zoomed in/out, OSX only updates the parts of Rects currently
displayed on screen, with relative location to current top-left on screen.
This will cause problems when we copy the dirty regions to the captured
frame. So we invalidate the whole screen to copy all the screen contents.

- With CGI method, the zooming will be ignored and the whole screen contents
will be captured as before.
- With IOSurface method, the zoomed screen contents will be captured.

Since we can't know the zooming level and focusing location, so we have
to copy the whole screen region for each frame during rooming. And this
will impact peformance a bit (with IOSurface capturer about 5-10 fps
down on MBP.)

Bug: chromium:911862
Change-Id: Icf123cde4d686ab7ce28fa731bc8dac6925492c8
Reviewed-on: https://webrtc-review.googlesource.com/c/113101
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25936}
2018-12-07 17:22:35 +00:00
4348ce240a Calculate min and max receive timestamps for packets in a video frame
Bug: webrtc:10106
Change-Id: I1d3469abb1e7bb7c91a5912d7b781505526abaca
Reviewed-on: https://webrtc-review.googlesource.com/c/113507
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25935}
2018-12-07 16:22:34 +00:00
48a79465ec Convert all webrtc code to not access EncodedImage::_size directly.
Read using capacity() method, write using set_buffer() method. This is
a preparation for making the member private, and renaming it to
capacity_.

Bug: webrtc:9378
Change-Id: I2f96679d052a83fe81be40301bd9863c87074640
Reviewed-on: https://webrtc-review.googlesource.com/c/113520
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25934}
2018-12-07 16:19:34 +00:00
60d770f2ef Replace luci-go dependency to CIPD package
This is needed to be compatible with chromium change, see bug for
details.

BUG=chromium:851596

Change-Id: I7b3ffda3715e925c42f4b95a2ba1d3f5cf829fda
Reviewed-on: https://webrtc-review.googlesource.com/c/113504
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25933}
2018-12-07 15:35:22 +00:00
10b051083c Disable hermetic toolchain when building on macOS 10.14.
This is copied from https://chromium-review.googlesource.com/1333877
More info is available at crbug.com/904400.

Bug: webrtc:10093
Change-Id: Ia256b3515b354b501663f0536c2735542474d3c0
Reviewed-on: https://webrtc-review.googlesource.com/c/113422
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25932}
2018-12-07 13:57:28 +00:00
3f10ca8145 Always record receive timestamps even on when the invalid flag is set.
This change is based on a discussion for integrating a new statistic that
measures the delay between the first frame being received and the first frame
being decoded. To enable this in the context of FrameEncryption it makes sense
for packet receive timestamps to be unconditionally recorded.

Bug: webrtc:10105
Change-Id: I6b3b0118121db1fe5d4a4fb16cf5d94341cd2b1b
Reviewed-on: https://webrtc-review.googlesource.com/c/113487
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25931}
2018-12-07 12:29:45 +00:00
d1d7b23f89 Include protection bitrate in total max allocated bitrate
This way we make sure we take fec into account when deciding how high
we probe.

Bug: webrtc:10070
Change-Id: I5286c82fc32dd99f7b9d79c9e5fc4465e1c6c259
Reviewed-on: https://webrtc-review.googlesource.com/c/113429
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25930}
2018-12-07 10:43:55 +00:00
120469086a Export the standardized IceConnectionState.
Since a lot of native users have taken dependencies on our old, non-standard behaviour
we'll have to have two ice connection states living side by side until we can get rid
of the old one.

Bug: webrtc:6145
Change-Id: I9b673bffeb1dfcf410f7c56d4def5912121e644c
Reviewed-on: https://webrtc-review.googlesource.com/c/113421
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25929}
2018-12-07 09:54:59 +00:00
6c95e2d56a Fuzz unfuzzed AEC3 killswitch field trials
Bug: webrtc:9413
Change-Id: Iccf861453c1c49c306ad18542074a792592491a9
Reviewed-on: https://webrtc-review.googlesource.com/c/113501
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25928}
2018-12-07 09:38:49 +00:00
b7180c09fc Replace RegisterExternalDecoder in NetEq test VerifyTimestampPropagation.
Bug: webrtc:10080
Change-Id: Ie93f130863115c2d288cfd9f3e273a9fbc982ed6
Reviewed-on: https://webrtc-review.googlesource.com/c/112904
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25927}
2018-12-07 09:28:47 +00:00
d644feb81f Make sure media transport is deleted before ICE.
This was regression due to
https://webrtc-review.googlesource.com/c/src/+/111920

which broke downstream projects. The break was not caught in
unit tests, because unit tests use loopback and fake media
transports that do not use ICE.

Bug: None
Change-Id: If95935afed430d62d5ff9a2ee01d8eaccadc198d
Reviewed-on: https://webrtc-review.googlesource.com/c/113440
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25926}
2018-12-06 18:11:28 +00:00
f73d40feca Add visibility for rtc_vp9_profile
This is necessary to access profiles from Chrome side.

Bug: webrtc:7925
Change-Id: I27d187afb56da715caf9f2ac8a6942778853542c
Reviewed-on: https://webrtc-review.googlesource.com/c/113100
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25925}
2018-12-06 16:46:51 +00:00
cdc5eb0de1 Replace VideoEncoderFactory::QueryVideoEncoder with VideoEncoder::GetEncoderInfo
Make implementation of VideoEncoderFactory::QueryVideoEncoder optional
until it is removed downstream and remove all implementations of it.

Bug: webrtc:10065
Change-Id: Ibb1f9612234e536651ce53f05ee048a5d172a41f
Reviewed-on: https://webrtc-review.googlesource.com/c/113065
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25924}
2018-12-06 15:24:45 +00:00
0b6d0e6e38 Introduce some placeholder build targets
So that users can add dependencies on them, and not break when a bunch
of headers move out of rtc_base:rtc_base.

Bug: webrtc:9987
Change-Id: Iecd5dd903cb8b97cb6f051e3a0cb6df7f8ba22b3
Reviewed-on: https://webrtc-review.googlesource.com/c/113425
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25923}
2018-12-06 15:09:44 +00:00
87609be863 Merges RtpTransportControllerSend with SendSideCongestionController.
Bug: webrtc:9586
Change-Id: I50332f2e128f107e40af7776be0ed530e20774d9
Reviewed-on: https://webrtc-review.googlesource.com/c/113183
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25922}
2018-12-06 13:38:39 +00:00
722875f72e Adding partial authentication of the Generic RTP Frame Descriptor.
Bug: None
Change-Id: I590e28acbd17b45dcb4e3bac34d223ad0903f7dc
Reviewed-on: https://webrtc-review.googlesource.com/c/113131
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25921}
2018-12-06 13:35:59 +00:00
e7862cc6b5 Copy VP8EncoderSimulcastProxy to EncoderSimulcastProxy
Use the new class internally where appropriate too.

The objective is to rename it, but due to some external dependency,
it is better to copy, update dependencies and remove.

Bug: webrtc:10069
Change-Id: I8477ce5a2982933db27513cc9509f51558dafaf3
Reviewed-on: https://webrtc-review.googlesource.com/c/113265
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25920}
2018-12-06 13:24:07 +00:00
18f0c3c038 Add RegisterAudioSendPayload() method
In preparation of removing CodecInst.

Bug: webrtc:7626
Change-Id: I8955d17dbb3ec15177e505ae420376b542d48410
Reviewed-on: https://webrtc-review.googlesource.com/c/113306
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25919}
2018-12-06 12:44:53 +00:00
d8a1b7a5c5 Use opaque int as payload_type in MediaTransportInterface
Replaces enum VideoCodecType for video frames and uint8_t for audio
frames.

Also delete method
MediaTransportVideoSinkInterface::OnKeyFrameRequested; it needs to be
added as a send-side interface instead (for a later cl).

Bug: webrtc:9719
Change-Id: I2cfdbacc267afc75c448512e2cc6de0ec9966a2d
Reviewed-on: https://webrtc-review.googlesource.com/c/113180
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25918}
2018-12-06 12:37:27 +00:00
5658ea660e Fix iOS version for internal.client.webrtc perf test
This adjusts iOS version to the actual one on the tester bot.

Bug: webrtc:10047
Change-Id: I7d104f331450192142c8c2c1259a3207dcee45ed
Reviewed-on: https://webrtc-review.googlesource.com/c/113420
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25917}
2018-12-06 12:26:47 +00:00
a728c919a4 Fix performance bug in CreateSendSideBweSimulationGraph
Bug: webrtc:10097
Change-Id: Ie60619084cd4bd47f5f81d06262ba62631eac12f
Reviewed-on: https://webrtc-review.googlesource.com/c/113423
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25916}
2018-12-06 12:16:36 +00:00
628f37a6fe Delete a cricket::DtlsTransport when PC is closed
This avoids use-after-free problems that occur when references
to webrtc::DtlsTransport objects are held outside of the PC.

Bug: chromium:907849
Change-Id: Id428c8e616482eff0f4327d2eac17e29bb3f6484
Reviewed-on: https://webrtc-review.googlesource.com/c/113303
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25915}
2018-12-06 12:06:34 +00:00
fafae11bfc Allow parsing RTC event logs without a transaction ID.
Bug: webrtc:9972
Change-Id: I01d60671d249adbd55f25c8f49f205b18787cbf4
Reviewed-on: https://webrtc-review.googlesource.com/c/113304
Reviewed-by: Zach Stein <zstein@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25914}
2018-12-06 09:59:45 +00:00
518add35a8 Roll chromium_revision 68c6b982a7..d6dec3971c (614157:614285)
Change log: 68c6b982a7..d6dec3971c
Full diff: 68c6b982a7..d6dec3971c

Changed dependencies
* src/base: 8778f36ffb..1bc039647f
* src/build: f8704051e5..84f0bf98ad
* src/ios: f2a06afe6e..11779ae7d1
* src/testing: 92b188b3ea..897a09fa69
* src/third_party: 03c7212525..6d9122ca49
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/fa0beb5bf0..c7cc237f95
* src/third_party/depot_tools: 19238fc343..2e00228777
* src/tools: 4589892399..1c79b0fc32
DEPS diff: 68c6b982a7..d6dec3971c/DEPS

No update to Clang.

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

Change-Id: Idac08c10b9de476dd5243e8de6e5cf8d883e5d1e
Reviewed-on: https://webrtc-review.googlesource.com/c/113383
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/master@{#25913}
2018-12-06 08:22:11 +00:00
5f93a3eaa9 Roll chromium_revision 48afe1fce1..68c6b982a7 (614046:614157)
Change log: 48afe1fce1..68c6b982a7
Full diff: 48afe1fce1..68c6b982a7

Changed dependencies
* src/base: 38c52177ea..8778f36ffb
* src/ios: ffc1609b07..f2a06afe6e
* src/testing: 3f5136437f..92b188b3ea
* src/third_party: af23677231..03c7212525
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/58f298110d..fa0beb5bf0
* src/tools: 10a524fa51..4589892399
DEPS diff: 48afe1fce1..68c6b982a7/DEPS

No update to Clang.

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

Change-Id: Iefb2e2dd3376c7e13726e42003d234d02fbae214
Reviewed-on: https://webrtc-review.googlesource.com/c/113326
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/master@{#25912}
2018-12-05 23:37:41 +00:00
c60a77731d Process RTP before RTCP in RTC event log analyzer.
This handles an unlikely corner case where you receive a RTCP feedback for a packet the same millisecond that you send it.

Bug: None
Change-Id: I77f460bef4073d4d9c5633c88f4d2dd8470f8577
Reviewed-on: https://webrtc-review.googlesource.com/c/113305
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25911}
2018-12-05 21:04:55 +00:00
75726f2852 Roll chromium_revision 92bac9a669..48afe1fce1 (613922:614046)
Change log: 92bac9a669..48afe1fce1
Full diff: 92bac9a669..48afe1fce1

Changed dependencies
* src/base: 23fe4e6235..38c52177ea
* src/ios: b945749f3b..ffc1609b07
* src/testing: a5684e641c..3f5136437f
* src/third_party: cc7029ba2b..af23677231
* src/third_party/depot_tools: 687ca907fd..19238fc343
* src/third_party/harfbuzz-ng/src: 000d4b128e..79e7e3445e
* src/third_party/r8: ndmKWh0vZhDc2iLXEETOuWXVfafHbqwI_FcSgJJIfpoC..gMAAlElX8RMw__5KOpk-Ckdx3XDyEXspJVslmnblsrgC
* src/tools: cf5c7c5fbd..10a524fa51
DEPS diff: 92bac9a669..48afe1fce1/DEPS

No update to Clang.

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

Change-Id: Ibce718cf581f1626398c0db6620e8e15f8cd7041
Reviewed-on: https://webrtc-review.googlesource.com/c/113322
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/master@{#25910}
2018-12-05 19:41:15 +00:00
4a817f6e6f Store timestamp of first and last event in RTC event log parser.
This was previously only done for the legacy wire format.

Bug: webrtc:8111
Change-Id: I82767c8eafb35a50967c6ff3d0131c3981957c1f
Reviewed-on: https://webrtc-review.googlesource.com/c/112590
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25909}
2018-12-05 17:05:43 +00:00
bd2cf71865 Remove functions to inject a TaskQueue in RtcEventLog creation.
The event log implementation will be simpler if it creates its own TaskQueue.
If we really need the "injectable" functionality, it could be achieved via a
TaskQueueFactory that returns a move-constructible TaskQueue.

Bug: webrtc:10085
Change-Id: I538be3dd77c09be2f5bae015227067acd6af8355
Reviewed-on: https://webrtc-review.googlesource.com/c/113140
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25908}
2018-12-05 16:35:04 +00:00
b438b5a33d Reland "Change ReceiveStatistics reaction to large sequence numbers jumps"
This reverts commit 7e0299e2452b021fcd14a8fdb86257459eeacf90.

Reason for revert: audio receive stream fix not to use 0 reordering threshold

Original change's description:
> Revert "Change ReceiveStatistics reaction to large sequence numbers jumps"
> 
> This reverts commit c4f120130f495e9726bf221356642de69125f4a2.
> 
> Reason for revert: breaks downstream tests due to zero max reordering for audio receive channels
> 
> Original change's description:
> > Change ReceiveStatistics reaction to large sequence numbers jumps
> > 
> > Consider stream restart when two sequential packets arrived far from
> > previous packets' sequence numbers.
> > instead of resetting on single one.
> > For packet loss calculation ignore sequence number gap during reset.
> > 
> > Bug: webrtc:9445, b/38179459
> > Change-Id: I0c2717ef8f9ec182b280ae757b5582f56d9afcef
> > Reviewed-on: https://webrtc-review.googlesource.com/c/111962
> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#25890}
> 
> TBR=danilchap@webrtc.org,asapersson@webrtc.org
> 
> Change-Id: Icc9f4d86d9f0b07f0fa2f3d443f9a90aa91f5e21
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9445, b/38179459
> Reviewed-on: https://webrtc-review.googlesource.com/c/113067
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25897}

TBR=danilchap@webrtc.org,asapersson@webrtc.org

Change-Id: I8747aa5cb6209b92fafefed077bc19d305d11db6
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9445, b/38179459
Reviewed-on: https://webrtc-review.googlesource.com/c/113263
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25907}
2018-12-05 16:31:00 +00:00
dc107965bd Fix AGC2 fuzzer coverage.
Bug: webrtc:10084
Change-Id: Icc51994fe5ab16188c41452e887cbe7a6b8b9aff
Reviewed-on: https://webrtc-review.googlesource.com/c/112941
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25906}
2018-12-05 15:55:42 +00:00
2a977cf466 For audio receive channel use default max reordering threshold instead of 0
setting max reordering recently has been fix to actually set it.
(https://webrtc-review.googlesource.com/c/src/+/111752)
Another recent change fix stats to skip counting large sequence number jumps as packet loss
(https://webrtc-review.googlesource.com/c/src/+/111962)

max reordering thresholds affects how packet loss is calculated.
Packet loss is then reported to remote sending participant in rtcp receiver reports.
Sender uses packet loss mostly for stats, but also e.g. for opus fec adjustment.

Setting threshold to zero de-facto imply all packets should be considered in order.
That bug was mitigated by two other bugs mentioned above

This change increase threshold to default 50 packets aligning it with Video receiver
and unblocks (re)landing 2nd fix

Bug: b/120482366
Change-Id: Iadda0c2148ed84dd83c01183cfe9285568db4e29
Reviewed-on: https://webrtc-review.googlesource.com/c/113064
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25905}
2018-12-05 14:52:23 +00:00
c979c46bda Re-add MSVC debug bots to CQ
This reverts commit e80e0132c165b604e7a6684b356cddd857a60f7e.

Compilation has been fixed in https://chromium-review.googlesource.com/c/chromium/src/+/1297536

No-Try: True
Bug: webrtc:9695, webrtc:10071
Change-Id: I0e5b657c9197f1b71c2473aae366f06b0daad3f5
Reviewed-on: https://webrtc-review.googlesource.com/c/113220
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25904}
2018-12-05 12:20:56 +00:00
b819ed6df0 Roll chromium_revision 2e285ebae2..92bac9a669 (613019:613922)
Change log: 2e285ebae2..92bac9a669
Full diff: 2e285ebae2..92bac9a669

Changed dependencies
* src/base: 62febbdbd7..23fe4e6235
* src/build: 8b1ff06550..f8704051e5
* src/ios: 2c8e8f83db..b945749f3b
* src/testing: da3cc6c84a..a5684e641c
* src/third_party: a862efe9b4..cc7029ba2b
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/6f862e54f2..58f298110d
* src/third_party/depot_tools: 0b287c5bca..687ca907fd
* src/third_party/harfbuzz-ng/src: e0307de818..000d4b128e
* src/tools: cc443eb2fd..cf5c7c5fbd
DEPS diff: 2e285ebae2..92bac9a669/DEPS

No update to Clang.

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

Change-Id: Ia5d418b0ca23b47f4ef34b163cf32a429e061711
Reviewed-on: https://webrtc-review.googlesource.com/c/113162
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25903}
2018-12-05 10:44:17 +00:00
657b296ff5 Reland "Remove CodecInst pt.1"
This is a reland of 056f9738bf7a3d16da45398239656e165c4e0851

Original change's description:
> Remove CodecInst pt.1
> 
> Update audio_coding tests to not use CodecInst.
> 
> Bug: webrtc:7626
> Change-Id: I880fb8d72d7d0a915d274e67feb6106f023697c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/112594
> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25879}

Bug: webrtc:7626
Change-Id: I5d6ca0baf6230bfe9bf95c2c25496d2a56812d90
Reviewed-on: https://webrtc-review.googlesource.com/c/112942
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25902}
2018-12-05 10:38:23 +00:00
5b1477839d [Unified Plan] If "a=msid" is missing, create default stream.
Prior to this CL, if the "a=msid" attribute was missing it was treated
the same as if "no streams" were explicitly signaled (a=msid:-); the
receivers would not be associated with any streams.

In order to support legacy endpoints that don't recognize "a=msid" that
assume the Plan B behavior of a stream being created anyway, this CL
creates a stream with a random ID in such cases. For background, see
https://github.com/web-platform-tests/wpt/pull/14054.

Bug: chromium:907508
Change-Id: I9d9dd0e4ba8f9941f8652f4d7873adc560777cd9
Reviewed-on: https://webrtc-review.googlesource.com/c/112900
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25901}
2018-12-05 09:53:21 +00:00
e5e36ddc40 Roll chromium_revision 3546854f59..2e285ebae2 (612694:613019) + fix JNI
This changelist is based on Chromium autoroller CL
https://webrtc-review.googlesource.com/c/src/+/112847
with additional JNI fixes needed to propagate upstream changes
introduced in
c99e905516


Change log: 3546854f59..2e285ebae2
Full diff: 3546854f59..2e285ebae2

Changed dependencies
* src/base: 0551460b2b..62febbdbd7
* src/build: 59f4bb0792..8b1ff06550
* src/ios: 0c78d113b3..2c8e8f83db
* src/testing: d387a4a97a..da3cc6c84a
* src/third_party: e31ab38349..a862efe9b4
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/1b98245e3c..6f862e54f2
* src/third_party/depot_tools: 016601cc21..0b287c5bca
* src/third_party/r8: uM1IGlYVeBYwmhwRCSMVqRvmu4YFlL7M2yLwZ1DWUvAC..ndmKWh0vZhDc2iLXEETOuWXVfafHbqwI_FcSgJJIfpoC
* src/tools: 476768d37c..cc443eb2fd
DEPS diff: 3546854f59..2e285ebae2/DEPS

No update to Clang.

No-Try: True
Bug: chromium:898660
Change-Id: I8be89e16d9639d96fc09f053e29414381a486846
Reviewed-on: https://webrtc-review.googlesource.com/c/112595
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25900}
2018-12-05 09:48:51 +00:00
04744aee2f Set priority of iOS test tasks to 30
The default Swarming priority is 200 but it's recommended to raise it.
Chrome's tasks are set to 30, and that can cause our tasks to be discarded.

Bug: chromium:911787
Change-Id: Ied5eed4bc37890ede6c29d2fd743e102f5622d11
Reviewed-on: https://webrtc-review.googlesource.com/c/113145
Reviewed-by: Artem Titarenko <artit@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25899}
2018-12-05 08:59:44 +00:00