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Author SHA1 Message Date
Sebastian Jansson
acab559c7b Adds overuse predictor to GoogCC. 
Bug: webrtc:10498
Change-Id: Ic97c16d28cbc1e30609f6c1daa3a61423d44641c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136924
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28012}
 2019-05-21 16:50:39 +00:00
Anton Sukhanov
c701dec22b Add GetTransportParametersOffer method for DatagramTransportInterface 
This change adds missing GetTransportParametersOffer, which is required for datagram transport setup. We have exactly the same method in MediaTransportInterface. It's possible to add a separate interface, which will be used in both Media and Datagram transports, but I do not want to overcomplicate it now until we know more about future of media and datagram transports.


Bug: webrtc:9719
Change-Id: I8b6c9ebc9522acba75f74da2e18e4bb1eb0d1e4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137861
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28011}
 2019-05-21 16:13:43 +00:00
Niels Möller
04a3cc1ad9 Delete rtc_base/unittest_main.cc 
Usage replaced with test/test_main.cc.

Bug: webrtc:5996
Change-Id: I65e7539f2072fb45255a3c1af0b10dd06e1701ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137805
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28010}
 2019-05-21 14:44:11 +00:00
Minyue Li
d703cd022f Revert "Avoid encrypting empty audio packet." 
This reverts commit b0ac94307e1787f83de2b9a2dc3b58309ea8654b.

Reason for revert: failing upstream tests

Original change's description:
> Avoid encrypting empty audio packet.
> 
> Bug: b/132861665
> Change-Id: I161ba8697ae88857927f27fa6d3914b7201fdeab
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137049
> Commit-Queue: Minyue Li <minyue@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28006}

TBR=brandtr@webrtc.org,kwiberg@webrtc.org,minyue@webrtc.org

Change-Id: I856436ad78bcc5310810283bb5547070781d0684
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: b/132861665
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137518
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28009}
 2019-05-21 14:13:52 +00:00
Sergey Silkin
19da5ced24 Formatting of WebRTC-Vp9InterLayerPred field trial. 
Use conventional style ../{Default|Disabled|Enabled} with parameter
inter_layer_pred_mode:{off|on|onkeypic} which maps directly to
InterLayerPredMode enum.

Bug: chromium:949536
Change-Id: If34e789b031d0db3eb2748b0b824492237ad5187
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137800
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28008}
 2019-05-21 13:09:09 +00:00
Fredrik Hernqvist
3be9da37bb Make unpack_aecdump unpack RuntimeSettings 
When running unpack_aecdump --full, unpack RuntimeSettings into files, on the format that can be imported into Audacity.
Output one file for each RuntimeSetting present in the aecdump. If outputting several WAV files, output file for each WAV file with corresponding time stamps.

Bug: webrtc:10643
Change-Id: If147e509d36207f5f838457354e2451df65549d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137426
Commit-Queue: Fredrik Hernqvist <fhernqvist@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28007}
 2019-05-21 12:38:15 +00:00
Minyue Li
b0ac94307e Avoid encrypting empty audio packet. 
Bug: b/132861665
Change-Id: I161ba8697ae88857927f27fa6d3914b7201fdeab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137049
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28006}
 2019-05-21 11:14:10 +00:00
Artem Titov
4d29ef063c Add periodic alive message logging to prevent test infra think, that test is dead 
Bug: webrtc:10138
Change-Id: Ib39ff6df81776a7784687be2dc16ab81c500cc3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137428
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28005}
 2019-05-21 11:09:18 +00:00
Harald Alvestrand
97716c0132 Implement max-channels for SCTP datachannels. 
This involves catching another callback from usrsctp.
It also moves the definition of "connected" a little later
in the sequence: From "ready to send data" to the reception
of the SCTP_COMM_UP event.

Bug: chromium:943976
Change-Id: Ib9e1b17d0cc356f19cdfa675159b29bf1efdcb55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137435
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28004}
 2019-05-21 10:24:41 +00:00
Sebastian Jansson
8abcf83b4f Adds IsEmpty to SampleStats. 
Bug: webrtc:9883
Change-Id: Ie8ef801cb60fd74c0354ff9fbbdbc33b7d105317
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137514
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28003}
 2019-05-21 09:41:41 +00:00
Danil Chapovalov
aaa114368e Use single argument PeerConnectionFactory factory in objc code 
Bug: webrtc:10284
Change-Id: If656af94636731d1caa208db78e460740edbf83c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137422
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28002}
 2019-05-21 08:20:04 +00:00
Niels Moller
9d1840c3df Revert "Delete NO_MAIN_THREAD_WRAPPING preprocessor define." 
This reverts commit 0f78c6b28dbc0c9caa555ce89ce91b0f08c510ea.

Reason for revert: Breaks downstream tests.

Original change's description:
> Delete NO_MAIN_THREAD_WRAPPING preprocessor define.
> 
> Since many tests rely on rtc::Thread::Current(), add an
> explicit rtc::AutoThread in the main() function used by tests.
> 
> Bug: webrtc:9714
> Change-Id: Id82121967c9621fe1c2945846009c48139fa57da
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/39680
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28000}

TBR=kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org

Change-Id: Iff939bb0d5ad0ea01b953321993733bb56c9070b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9714
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137512
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28001}
 2019-05-21 07:26:54 +00:00
Niels Möller
0f78c6b28d Delete NO_MAIN_THREAD_WRAPPING preprocessor define. 
Since many tests rely on rtc::Thread::Current(), add an
explicit rtc::AutoThread in the main() function used by tests.

Bug: webrtc:9714
Change-Id: Id82121967c9621fe1c2945846009c48139fa57da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/39680
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28000}
 2019-05-21 06:53:54 +00:00
chromium-webrtc-autoroll
e8602067db Roll chromium_revision cc9f0ad182..7a39eea5d8 (661517:661628) 
Change log: cc9f0ad182..7a39eea5d8
Full diff: cc9f0ad182..7a39eea5d8

Changed dependencies
* src/base: 0556ba3715..1d4c19a8a6
* src/build: 214debc9e9..12e7bf6a6d
* src/testing: 1781c1c8f4..6d481142ef
* src/third_party: bd0441c427..aa6915457b
* src/third_party/depot_tools: ad39f9d8f8..5716400ae2
* src/third_party/freetype/src: 2f4b740ce4..fbbcf50367
* src/tools: f944291000..ccc725a068
DEPS diff: cc9f0ad182..7a39eea5d8/DEPS

No update to Clang.

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

Change-Id: I63c2d001f4d4bfd65cf59506ee3ef3732e010d5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137940
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@{#27999}
 2019-05-21 06:40:55 +00:00
Karl Wiberg
053c371552 Audio coding: Don't choke when RTP timestamp rate > sample rate 
Bug: webrtc:10631
Change-Id: If0422786172502f039acc2cac5e8c13b637af54c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137048
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27998}
 2019-05-21 03:10:49 +00:00
chromium-webrtc-autoroll
d9f02f64e8 Roll chromium_revision e7b2a8fc98..cc9f0ad182 (661399:661517) 
Change log: e7b2a8fc98..cc9f0ad182
Full diff: e7b2a8fc98..cc9f0ad182

Changed dependencies
* src/base: 2264d66e4e..0556ba3715
* src/build: d29d3d06ce..214debc9e9
* src/ios: 4a1d64ef98..2cace45200
* src/testing: 0041123f74..1781c1c8f4
* src/third_party: d60635378e..bd0441c427
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/4d85003327..5655d8f9f1
* src/third_party/depot_tools: 7639f1999a..ad39f9d8f8
* src/third_party/freetype/src: 31757f969f..2f4b740ce4
* src/tools: 2f5568b74b..f944291000
DEPS diff: e7b2a8fc98..cc9f0ad182/DEPS

No update to Clang.

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

Change-Id: I1ebf2fd25b5dea871e5ac7f1025e58b3fe7d2bab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137840
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@{#27997}
 2019-05-20 23:39:13 +00:00
Anton Sukhanov
762076b886 Add flag to use datagram transport (without implementation) 
Integration with datagram transport will come in next CLs.

NOTE that since we now have implemented negotiation for media transport, we can replace configuration flags with field trials, but it will be done later for both media and datagram transports.

Bug: webrtc:9719
Change-Id: Icf062d030899d53d5646977ba195d1634050704b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137820
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27996}
 2019-05-20 22:42:31 +00:00
chromium-webrtc-autoroll
9f864be38a Roll chromium_revision f5d370078e..e7b2a8fc98 (660984:661399) 
Change log: f5d370078e..e7b2a8fc98
Full diff: f5d370078e..e7b2a8fc98

Changed dependencies
* src/base: 73710be437..2264d66e4e
* src/build: effe4569a4..d29d3d06ce
* src/buildtools: 1f329a6e26..9ea486bd06
* src/buildtools/linux64: git_revision:64b846c96daeb3eaf08e26d8a84d8451c6cb712b..git_revision:81ee1967d3fcbc829bac1c005c3da59739c88df9
* src/buildtools/mac: git_revision:64b846c96daeb3eaf08e26d8a84d8451c6cb712b..git_revision:81ee1967d3fcbc829bac1c005c3da59739c88df9
* src/buildtools/win: git_revision:64b846c96daeb3eaf08e26d8a84d8451c6cb712b..git_revision:81ee1967d3fcbc829bac1c005c3da59739c88df9
* src/ios: a873bd4962..4a1d64ef98
* src/testing: 8ea54a3a60..0041123f74
* src/third_party: fa0c76c94c..d60635378e
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/6ea34ccba4..4d85003327
* src/third_party/depot_tools: d7e41546c0..7639f1999a
* src/tools: 8b09ac4817..2f5568b74b
DEPS diff: f5d370078e..e7b2a8fc98/DEPS

Clang version changed 360094:361104
Details: f5d370078e..e7b2a8fc98/tools/clang/scripts/update.py

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

Change-Id: I30c0b2c0494139c089eef1e81662f7ad48b93cde
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137777
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@{#27995}
 2019-05-20 19:12:07 +00:00
Sebastian Jansson
871ac42597 Refactor of GoogCC debug printer. 
Simplifying the code to better fit with how it is used.

Bug: webrtc:9883
Change-Id: I2bd52f26b829413e516dee4f551cf36574275019
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136681
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27994}
 2019-05-20 18:40:26 +00:00
Sebastian Jansson
f4e085a499 Using absl traits for checks and logging. 
Bug: webrtc:9883
Change-Id: If4af810c1ba64c6c022c0fb5328a75527bec5934
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133622
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27993}
 2019-05-20 18:39:12 +00:00
Guido Urdaneta
1ff16c87aa Add RtpSenderInterface.SetStreams 
This is a reland of df5731e44d510e9f23a35b77e9e102eb41919bf4 with fixes
to avoid existing chromium tests to fail.

Instead of replacing the existing RtpSender::set_stream_ids() to
also fire OnRenegotiationNeeded(), this CL keeps the old
set_stream_ids() and adds the new RtpSender::SetStreams() which sets
the stream IDs and fires the callback.

This allows existing callsites to maintain behavior, and reserve
SetStreams() for the cases when we want OnRenegotiationNeeded() to fire.

Using the SetStreams() name instead of SetStreamIDs() to match the W3C
spec and to make it more different that the existing set_stream_ids().

Original change's description:
> Improve spec compliance of SetStreamIDs in RtpSenderInterface
>
> This CL makes RtpSender::SetStreamIDs fire fire negotiationneeded
> event if needed and exposes the method on RtpSenderInterface.
>
> This is a spec-compliance change.
>
> Bug: webrtc:10129
> Change-Id: I2b98b92665c847102838b094516a79b24de0e47e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135121
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27974}

Bug: webrtc:10129
Change-Id: Ic0b322bfa25c157e3a39465ef8b486f898eaf6bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137439
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27992}
 2019-05-20 18:38:06 +00:00
Karl Wiberg
eb16697259 AudioEncoderOpus: Don't mix up sample rate and RTP timestamp rate 
A later change will allow them to differ.

Bug: webrtc:10631
Change-Id: I4e13f41980261990b3bbbc6897cd754369265ca0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137046
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27991}
 2019-05-20 17:33:56 +00:00
Magnus Jedvert
94079f8452 Android: Add support for OpenGL ES 3 
Bug: webrtc:10642
Change-Id: I736e9e2520b364a817228a6599f4008d58165622
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137424
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27990}
 2019-05-20 16:39:56 +00:00
Alessio Bazzica
b9979a533c AGC2 RNN VAD: remove unused dep (KissFFT) 
NOTRY=True

Bug: webrtc:9577,webrtc:10480,webrtc:9139
Change-Id: I9fdf8c3bfd91d11fe01860546bcb83a78f5443fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137434
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27989}
 2019-05-20 14:43:33 +00:00
Henrik Andreassson
cc189177a6 Revert "Improve spec compliance of SetStreamIDs in RtpSenderInterface" 
This reverts commit df5731e44d510e9f23a35b77e9e102eb41919bf4.

Reason for revert: Breaks WebRTC in Chrome FYI for all platforms.

https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Mac%20Tester/2966

Original change's description:
> Improve spec compliance of SetStreamIDs in RtpSenderInterface
>
> This CL makes RtpSender::SetStreamIDs fire fire negotiationneeded
> event if needed and exposes the method on RtpSenderInterface.
>
> This is a spec-compliance change.
>
> Bug: webrtc:10129
> Change-Id: I2b98b92665c847102838b094516a79b24de0e47e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135121
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27974}

TBR=steveanton@webrtc.org,hbos@webrtc.org,guidou@webrtc.org

# Passing all bots except for flaky webrtc_perf_tests
NOTRY=True

Bug: webrtc:10129
Change-Id: If97317f7a01b34465685fcebbeea0d7576ed7328
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137431
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27988}
 2019-05-20 14:28:37 +00:00
Mirko Bonadei
39f46810ff Remove unused dependency. 
Bug: None
Change-Id: I13ef76d9f8410bda3591c5fc8a9607c768c92b65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137432
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27987}
 2019-05-20 12:57:44 +00:00
Mirko Bonadei
0ee0d1e15c Roll chromium_revision 243a2094e7..f5d370078e (660868:660984) 
Change log: 243a2094e7..f5d370078e
Full diff: 243a2094e7..f5d370078e

Changed dependencies
* src/base: fba03dece9..73710be437
* src/build: 3c7a12c795..effe4569a4
* src/ios: 11b06981d9..a873bd4962
* src/testing: 904b090729..8ea54a3a60
* src/third_party: fb42db204b..fa0c76c94c
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/b1d937f421..6ea34ccba4
* src/third_party/depot_tools: 0e405d1ac6..d7e41546c0
* src/third_party/r8: -hqyjKgjGWSfNsdcPJAnYNVOb96JOv0pJM82vtRo9M8C..jfE9VkwFvzhAgaBwY40d5HnUk_gcPl8H5vqsTQtb7DYC
* src/third_party/robolectric: iRFT1e5YFmRn5cbV0cAkQ5vDUXFmQ4qPYqStmmDfiMMC..1KXoOiNP1a_uZNdM2ybWKwAQNow1dHTXTig-ZK4Xgq8C
* src/tools: 912a00ef8d..8b09ac4817
DEPS diff: 243a2094e7..f5d370078e/DEPS

No update to Clang.

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

Change-Id: I1bb290e81d0c156b266d84d59e2943a7625fa2e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137484
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27986}
 2019-05-20 12:44:24 +00:00
Danil Chapovalov
03b4f9d1f8 Update android tests to use single argument PeerConnectionFactory factory 
Bug: webrtc:10284
Change-Id: Ifd3e2322f6fe01ed7ad9254c7d4e8cddca59b491
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137051
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27985}
 2019-05-20 12:27:32 +00:00
Niels Möller
60f4e29259 Delete configuration of unused transport_sequence_number_allocator 
RtpVideoStreamReceiver used to pass the PacketRouter when creating its
RtpRtcp module, but it's not needed for a receive-only module. Make the
PacketRouter optional to the constructor; it's used only for registering
the created RtpRtcp module as a candidate for sending rtcp feedback.

Bug: None
Change-Id: I371a0bdb9d68ac48b16f52e1d7939f8c177dc528
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137429
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27984}
 2019-05-20 12:26:27 +00:00
Artem Titov
ed4d1584cb Fix test names in pc_full_stack_tests.cc 
Bug: webrtc:10138
Change-Id: Id2ab0bd30fe3b80fa3fc3891d93e8ad6484d46e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137508
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27983}
 2019-05-20 12:25:23 +00:00
Niels Möller
5a96a0e516 Reland "Delete deprecated rtc::Thread default constructor" 
This is a reland of fdd6d3e46e22e1242aa4acd7aa0271a7562fb0ac

Original change's description:
> Delete deprecated rtc::Thread default constructor
>
> Bug: None
> Change-Id: Ic0e2e94b174a49e5d20ebdea90568473e1b71d62
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134640
> Reviewed-by: Tommi <tommi@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27958}

Tbr: tommi@webrtc.org
Bug: None
Change-Id: I9e4b1d06e79670b4efb9c9517d909a0562485e12
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137504
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27982}
 2019-05-20 07:49:30 +00:00
Artem Titov
137f6c8952 Introduce peer connection level webrtc video quality tests. 
Add video quality tests on new PC level framework basing on
full_stack_test.cc.

Bug: webrtc:10138
Change-Id: Id669e9022d5a111512978b6f69dbe2013eb43c8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136802
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27981}
 2019-05-18 11:24:25 +00:00
Anton Sukhanov
6b319e68e9 Add CreateDatagram to MediaTransportFactory 
Bug: webrtc:9719
Change-Id: I6e756d925917f032aa94a221706cd4241085b2a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137340
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27980}
 2019-05-17 22:18:41 +00:00
Henrik Boström
9fe1834d5d Implement RTCOutboundRtpStreamStats.totalPacketSendDelay for video. 
This is a standardized metric. Spec:
https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalpacketsenddelay

It is meant to replace the legacy googBucketDelay. The average
packet delay over any interval can be calculated as the delta
totalPacketSendDelay divided by the delta packetsSent between two
calls to getStats().

Bug: webrtc:10506
Change-Id: I3d6c6d66e5a06937d0ea8d182a82cd255084ad19
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137044
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27979}
 2019-05-17 18:53:20 +00:00
Tom Anderson
45b2e27ccd Remove non-source sources from binary targets 
No behavior changes.

BUG=chromium:964411

Change-Id: I833cfc4571d2f191acbb53e4b423f2a174ac9de3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137520
Commit-Queue: Thomas Anderson <thomasanderson@chromium.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27978}
 2019-05-17 18:43:00 +00:00
Anton Sukhanov
b50d995a5b Add juberti@ to webrtc root owners 
It's useful to have someone in PST timezone with root approval.

Bug: webrtc:10638
Change-Id: I9453c6bca3b77879c0c893bff8ca1abb6db5ab2a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137182
Reviewed-by: Justin Uberti <juberti@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27977}
 2019-05-17 18:11:58 +00:00
chromium-webrtc-autoroll
12f886640f Roll chromium_revision 4c9872694a..243a2094e7 (660753:660868) 
Change log: 4c9872694a..243a2094e7
Full diff: 4c9872694a..243a2094e7

Changed dependencies
* src/base: 5a3c8a6d57..fba03dece9
* src/build: 865cb800d6..3c7a12c795
* src/ios: dd4258e733..11b06981d9
* src/testing: 9d9a16dab9..904b090729
* src/third_party: 472de41bf6..fb42db204b
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/fd64d5d2d4..b1d937f421
* src/third_party/depot_tools: 5737f025b5..0e405d1ac6
* src/tools: 8c467d7632..912a00ef8d
DEPS diff: 4c9872694a..243a2094e7/DEPS

No update to Clang.

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

Change-Id: I8d7631c35d9a83c9b9e137d1e020117f8aeb42da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137481
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@{#27976}
 2019-05-17 17:31:46 +00:00
Sebastian Jansson
3525f86c42 Adds feedback generator. 
This is a useful tool to use for unittests of code that uses
TransportFeedback as input.

Bug: webrtc:10498
Change-Id: I171b22841eb9e16a5d5b785ff45ae9df5a6ccd7f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137423
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27975}
 2019-05-17 16:14:32 +00:00
Guido Urdaneta
df5731e44d Improve spec compliance of SetStreamIDs in RtpSenderInterface 
This CL makes RtpSender::SetStreamIDs fire fire negotiationneeded
event if needed and exposes the method on RtpSenderInterface.

This is a spec-compliance change.

Bug: webrtc:10129
Change-Id: I2b98b92665c847102838b094516a79b24de0e47e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135121
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27974}
 2019-05-17 12:53:31 +00:00
Artem Titov
519d74a5fc Drop data for disabled endpoints. 
Drop packets received from disabled endpoint and return socket error
when trying to send data from disabled endpoint.

Bug: webrtc:10138
Change-Id: I55259d2ac47adea78b47aeb25842e63a98a405c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134643
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27973}
 2019-05-17 10:31:22 +00:00
Niels Möller
fd26ef732f Delete unused RTPFragmentationHeader members 
Deleted fragmentationTimeDiff and fragmentationPlType. Unused since cl
https://webrtc-review.googlesource.com/c/src/+/134212.

Bug: webrtc:6471
Change-Id: I36b45be6f6babeda5a5f172c1f1a3876bb752e7f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134308
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27972}
 2019-05-17 09:26:17 +00:00
Niels Möller
f13a0960e6 Fix memory leak in Thread::PostTask. 
Use MessageData rather than MessageHandler to refer
to allocated storage.

That way, MessageQueue will delete storage for us if the
thread object is stopped before the Message is handled.

Leak seems triggered by the
RTCStatsIntegrationTest.GetsStatsWhileClosingPeerConnection
test.

Bug: webrtc:9714
Change-Id: I9e1255a3b6f16a763568744775ec0b3aef671227
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136684
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27971}
 2019-05-17 08:45:32 +00:00
Ruslan Burakov
1e193faaf1 Add DecelerationTargetLevelOffset Field Trial. 
This change allows NetEq to reach preferred jitter buffer size much faster
for high target delays because it uses absolute units instead of relative ones
during computation of lower_limit.

More details can be found here:
https://docs.google.com/document/d/12qPMJYFhGXrA_o_nvz9VshpzAJX6aULxFig1fTzBzDI/edit

Change-Id: I21ce0e35e25166d935fdf0325c083bcf990899f5

Bug: webrtc:10619
Change-Id: I21ce0e35e25166d935fdf0325c083bcf990899f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135745
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Cr-Commit-Position: refs/heads/master@{#27970}
 2019-05-17 08:08:12 +00:00
chromium-webrtc-autoroll
8e1a0080d3 Roll chromium_revision 64564f7a42..4c9872694a (660541:660753) 
Change log: 64564f7a42..4c9872694a
Full diff: 64564f7a42..4c9872694a

Changed dependencies
* src/base: 79d0db3862..5a3c8a6d57
* src/build: 05ee3a4249..865cb800d6
* src/ios: 9efbd407c6..dd4258e733
* src/testing: f086dcc9cc..9d9a16dab9
* src/third_party: b57abf30b7..472de41bf6
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/72ee2533b1..fd64d5d2d4
* src/third_party/depot_tools: 99fe071354..5737f025b5
* src/tools: 3036709251..8c467d7632
DEPS diff: 64564f7a42..4c9872694a/DEPS

No update to Clang.

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

Change-Id: Ib8805f3343f87366d0e9b7a3aaa822fe82a7a5ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137405
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@{#27969}
 2019-05-17 07:44:07 +00:00
Mirko Bonadei
02ed529340 Revert "Roll chromium_revision 64564f7a42..5ab21b0a3a (660541:660661)" 
This reverts commit 45a827351bfc2406d4ef08eab9c551cdb8b93594.

Reason for revert: It contains https://chromium-review.googlesource.com/c/chromium/src/+/1610749 which breaks Android compilation at HEAD.

Original change's description:
> Roll chromium_revision 64564f7a42..5ab21b0a3a (660541:660661)
> 
> Change log: 64564f7a42..5ab21b0a3a
> Full diff: 64564f7a42..5ab21b0a3a
> 
> Changed dependencies
> * src/base: 79d0db3862..2af57fa3f5
> * src/build: 05ee3a4249..747602a080
> * src/ios: 9efbd407c6..8144c1a2dc
> * src/testing: f086dcc9cc..27aef0038a
> * src/third_party: b57abf30b7..78419dc6d6
> * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/72ee2533b1..fd64d5d2d4
> * src/tools: 3036709251..5d994ede45
> DEPS diff: 64564f7a42..5ab21b0a3a/DEPS
> 
> No update to Clang.
> 
> TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
> BUG=None
> 
> Change-Id: Ic4eca55ad692fc06a88d26b9ae40ddab68e8c384
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137320
> 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@{#27967}

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com

Change-Id: I9b1d68d02ab4fc8dd237245ce86b5a56c544637c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137420
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27968}
 2019-05-17 07:02:15 +00:00
chromium-webrtc-autoroll
45a827351b Roll chromium_revision 64564f7a42..5ab21b0a3a (660541:660661) 
Change log: 64564f7a42..5ab21b0a3a
Full diff: 64564f7a42..5ab21b0a3a

Changed dependencies
* src/base: 79d0db3862..2af57fa3f5
* src/build: 05ee3a4249..747602a080
* src/ios: 9efbd407c6..8144c1a2dc
* src/testing: f086dcc9cc..27aef0038a
* src/third_party: b57abf30b7..78419dc6d6
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/72ee2533b1..fd64d5d2d4
* src/tools: 3036709251..5d994ede45
DEPS diff: 64564f7a42..5ab21b0a3a/DEPS

No update to Clang.

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

Change-Id: Ic4eca55ad692fc06a88d26b9ae40ddab68e8c384
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137320
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@{#27967}
 2019-05-17 00:31:05 +00:00
chromium-webrtc-autoroll
fe57f6252f Roll chromium_revision 0d85f6ad4e..64564f7a42 (660414:660541) 
Change log: 0d85f6ad4e..64564f7a42
Full diff: 0d85f6ad4e..64564f7a42

Changed dependencies
* src/base: d2b96a427d..79d0db3862
* src/build: 2910da43de..05ee3a4249
* src/ios: 703ce24672..9efbd407c6
* src/testing: 3ea10302b0..f086dcc9cc
* src/third_party: 98b50663f6..b57abf30b7
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/89ec6c772d..72ee2533b1
* src/tools: 85b42ed358..3036709251
DEPS diff: 0d85f6ad4e..64564f7a42/DEPS

No update to Clang.

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

Change-Id: I4b3b3e7e5976e72a2a6e4647e90f4e25d46f506a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137240
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@{#27966}
 2019-05-16 20:26:24 +00:00
chromium-webrtc-autoroll
f0792ce410 Roll chromium_revision 609f581dc6..0d85f6ad4e (660301:660414) 
Change log: 609f581dc6..0d85f6ad4e
Full diff: 609f581dc6..0d85f6ad4e

Changed dependencies
* src/base: 18f051a0f0..d2b96a427d
* src/build: 333f8224f9..2910da43de
* src/ios: 394135d276..703ce24672
* src/third_party: c5eea7e75a..98b50663f6
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/d84db82ff3..89ec6c772d
* src/tools: 87a822c49d..85b42ed358
DEPS diff: 609f581dc6..0d85f6ad4e/DEPS

No update to Clang.

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

Change-Id: Ic4cbe4d2852aba0b4764b9881509159376b32cd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137140
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@{#27965}
 2019-05-16 16:34:29 +00:00
Mirta Dvornicic
a24d934ee4 Add the option to use raw RTP packetization without the generic header. 
Bug: webrtc:10625
Change-Id: I198031154dbb706ae1e7c15bd34a3bdf93d1a51a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136923
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27964}
 2019-05-16 14:41:42 +00:00
Per Åhgren
67c76b214d AEC3: Minor code corrections 
Bug: webrtc:8671
Change-Id: I096053087e7ef0f3375f9c20b55558c1cec670cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136806
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27963}
 2019-05-16 13:53:27 +00:00