Commit Graph

232 Commits

Author SHA1 Message Date
d15a575ec3 Use SequenceChecker from public API
Bug: webrtc:12419
Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 15:04:55 +00:00
2accc7d6e0 Revert "Add task queue to RtpRtcpInterface::Configuration."
This reverts commit f23e2144e86400e2d68097345d4b3dc7a4b7f8a4.

Reason for revert: Need further discussion on appropriate thread/tq requirements.

Original change's description:
> Add task queue to RtpRtcpInterface::Configuration.
>
> Let ModuleRtpRtcpImpl2 use the configured value instead of
> TaskQueueBase::Current().
>
> Intention is to allow construction of RtpRtcpImpl2 on any thread.
> If a task queue is provided (required for periodic rtt updates), the
> destruction of the object must be done on that same task queue.
>
> Also, delete ModuleRtpRtcpImpl2::Create, callers updated to use std::make_unique.
>
> Bug: None
> Change-Id: I412b7b1e1ce24722ffd23d16aa6c48a7214c9bcd
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/199968
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32949}

TBR=danilchap@webrtc.org,ilnik@webrtc.org,saza@webrtc.org,nisse@webrtc.org,srte@webrtc.org

Change-Id: I7e5007f524a39a6552973ec9744cd04c13162432
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201420
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32953}
2021-01-12 17:47:32 +00:00
f23e2144e8 Add task queue to RtpRtcpInterface::Configuration.
Let ModuleRtpRtcpImpl2 use the configured value instead of
TaskQueueBase::Current().

Intention is to allow construction of RtpRtcpImpl2 on any thread.
If a task queue is provided (required for periodic rtt updates), the
destruction of the object must be done on that same task queue.

Also, delete ModuleRtpRtcpImpl2::Create, callers updated to use std::make_unique.

Bug: None
Change-Id: I412b7b1e1ce24722ffd23d16aa6c48a7214c9bcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/199968
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32949}
2021-01-12 12:42:58 +00:00
36274f9158 Reland "Reland "Default enable WebRTC-SendSideBwe-WithOverhead.""
This is a reland of 1dbe30c7e895c7eb4da51c968a7a8897f25ad7e6

Original change's description:
> Reland "Default enable WebRTC-SendSideBwe-WithOverhead."
>
> This is a reland of 87c1950841c3f5e465e1663cc922717ce191e192
>
> Original change's description:
> > Default enable WebRTC-SendSideBwe-WithOverhead.
> >
> > Bug: webrtc:6762
> > Change-Id: I18ace06a33b3b60d5a19796d4769f70cd977d604
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188801
> > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Ali Tofigh <alito@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#32472}
>
> Bug: webrtc:6762
> Change-Id: Icf096a8755d29600a13bd08b1f22f5a79de21e90
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190143
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32492}

Bug: webrtc:6762
Change-Id: I6d79894a213fc42d2338409e7513247725881b1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191221
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32534}
2020-11-02 11:05:56 +00:00
d546186b89 Revert "Reland "Default enable WebRTC-SendSideBwe-WithOverhead.""
This reverts commit 1dbe30c7e895c7eb4da51c968a7a8897f25ad7e6.

Reason for revert: Speculative revert due to failing tests.

Original change's description:
> Reland "Default enable WebRTC-SendSideBwe-WithOverhead."
>
> This is a reland of 87c1950841c3f5e465e1663cc922717ce191e192
>
> Original change's description:
> > Default enable WebRTC-SendSideBwe-WithOverhead.
> >
> > Bug: webrtc:6762
> > Change-Id: I18ace06a33b3b60d5a19796d4769f70cd977d604
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188801
> > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Ali Tofigh <alito@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#32472}
>
> Bug: webrtc:6762
> Change-Id: Icf096a8755d29600a13bd08b1f22f5a79de21e90
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190143
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32492}

TBR=stefan@webrtc.org,jakobi@webrtc.org,alito@webrtc.org

Change-Id: I7e0378788576236059627cf8c3bad58cd70aff7e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6762
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190500
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32504}
2020-10-27 10:51:46 +00:00
1dbe30c7e8 Reland "Default enable WebRTC-SendSideBwe-WithOverhead."
This is a reland of 87c1950841c3f5e465e1663cc922717ce191e192

Original change's description:
> Default enable WebRTC-SendSideBwe-WithOverhead.
>
> Bug: webrtc:6762
> Change-Id: I18ace06a33b3b60d5a19796d4769f70cd977d604
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188801
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32472}

Bug: webrtc:6762
Change-Id: Icf096a8755d29600a13bd08b1f22f5a79de21e90
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190143
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32492}
2020-10-26 12:35:47 +00:00
9d69cbeabf Changes default pacing factor to 1.1x
This changes the default behavior to use pacing factor of 1.1x instead
of 2.5x, it also sets libvpx rate controler as trusted, turns on the
encoder pushback mechanism and sets spatial hysteresis to 1.2.
The unused "dynamic rate" settings in libvpx is removed.

The new settings matches what has been used in chromium since 2019.
If needed, the legacy behavior can be enabled using the field trial
WebRTC-VideoRateControl.

Bug: webrtc:10155
Change-Id: I8186b491aa5bef61e8f568e96c980ca68f0c208f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186661
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32477}
2020-10-23 13:43:32 +00:00
27af3c4c24 Revert "Default enable WebRTC-SendSideBwe-WithOverhead."
This reverts commit 87c1950841c3f5e465e1663cc922717ce191e192.

Reason for revert: breaks downstream tests

Original change's description:
> Default enable WebRTC-SendSideBwe-WithOverhead.
>
> Bug: webrtc:6762
> Change-Id: I18ace06a33b3b60d5a19796d4769f70cd977d604
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188801
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32472}

TBR=stefan@webrtc.org,jakobi@webrtc.org,alito@webrtc.org

Change-Id: If59fd41dcd8f6db76ea297c34c25fe19ae2ae973
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6762
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189973
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32474}
2020-10-22 16:57:18 +00:00
87c1950841 Default enable WebRTC-SendSideBwe-WithOverhead.
Bug: webrtc:6762
Change-Id: I18ace06a33b3b60d5a19796d4769f70cd977d604
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188801
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32472}
2020-10-22 13:37:18 +00:00
279f37052c Makes WebRTC-Pacer-SmallFirstProbePacket default enabled.
This is expected to yield slightly higher bandwidth estimates when
probing is used, since it reduces a bias in how packet sizes are counted.

Bug: webrtc:11780
Change-Id: I6a4a3af0c50670d248dbe043a4d9da60915e3699
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187491
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32394}
2020-10-13 21:45:42 +00:00
3c65a2b5d4 Cleans up code related to video packetization overhead.
Bug: webrtc:10155
Change-Id: Idc3aaa916e38a18e5ca9dc861587e96465915c64
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186660
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32312}
2020-10-05 13:22:33 +00:00
12f465cf94 Deprecate the raw-pointer constructor of EncodedImage.
Bug: webrtc:9378
Change-Id: I5591202aff3e9f22e902f52096ddb0592662789e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185008
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32175}
2020-09-23 12:51:30 +00:00
2549f174b5 Remove RTPFragmentationHeader creation and propagation through webrtc
Bug: webrtc:6471
Change-Id: I5cb1e10088aaecb5981888082b87ae9957bbaaef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181541
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31955}
2020-08-17 16:37:33 +00:00
fca30875b6 Do not use internal source in video send stream tests.
Unless testing a behavior specific to video encoders with internal source.

Bug: None
Change-Id: Ia42177f2a0b7e658060208444870a7466c6779dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179362
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31740}
2020-07-16 09:42:47 +00:00
a376518817 Migrate video/ except video/end_to_end_tests and video/adaptation to webrtc::Mutex.
Also migrates test/ partly.

Bug: webrtc:11567
Change-Id: If5b2eae65c5f297f364b6e3c67f94946a09b4a96
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178862
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31672}
2020-07-08 12:21:08 +00:00
a827a30bb7 Revert "Migrate video/ except video/end_to_end_tests and video/adaptation to webrtc::Mutex."
This reverts commit 0eba415fb40cc4e3958546a8ee53c698940df0a1.

Reason for revert: previously unknown lock recursion occurring downstream.

Original change's description:
> Migrate video/ except video/end_to_end_tests and video/adaptation to webrtc::Mutex.
> 
> Also migrates test/ partly.
> 
> Bug: webrtc:11567
> Change-Id: I4203919615c087e5faca3b2fa1d54cba9f171e07
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178813
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31653}

TBR=sprang@webrtc.org,handellm@webrtc.org

Change-Id: I13f337e0de5b8f0eb19deb57cb5623444460ec4d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11567
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178842
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31656}
2020-07-07 20:46:48 +00:00
0eba415fb4 Migrate video/ except video/end_to_end_tests and video/adaptation to webrtc::Mutex.
Also migrates test/ partly.

Bug: webrtc:11567
Change-Id: I4203919615c087e5faca3b2fa1d54cba9f171e07
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178813
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31653}
2020-07-07 18:01:44 +00:00
f25761d798 Remove dependency from RtpRtcp on the Module interface.
The 'Module' part of the implementation must not be
called via the RtpRtcp interface, but is rather a part of
the contract with ProcessThread. That in turn is an
implementation detail for how timers are currently implemented
in the default implementation.

Along the way I'm deprecating away the factory function which
was inside the interface and tied it to one specific implementation.
Instead, I'm moving that to the implementation itself and down the
line, we don't have to go through it if we just want to create an
instance of the class.

The key change is in rtp_rtcp.h and the new rtp_rtcp_interface.h
header file (things moved from rtp_rtcp.h), the rest falls from that.

Change-Id: I294f13e947b9e3e4e649400ee94a11a81e8071ce
Bug: webrtc:11581
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176419
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31440}
2020-06-04 08:11:21 +00:00
fae05624ec Deprecate the static RtpRtcp::Create() method.
The method is being used externally to create instances
of the deprecated internal implementation.

Instead, I'm moving how we instantiate the internal implementation into
the implementation itself and move towards keeping the interface
separate from a single implementation.

Change-Id: I743aa86dc4c812b545699c546c253c104719260e
Bug: webrtc:11581
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176404
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31420}
2020-06-03 09:41:34 +00:00
980e0c16b9 VideoSendStreamTest: remove lock recursions.
This change removes lock recursions and adds thread annotations.

Bug: webrtc:11567
Change-Id: Id3f375ed925b0b8f3cad449d0971d9236088b1d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176225
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31382}
2020-05-29 07:55:06 +00:00
3a5742c880 Add thread/sequence checks to ModuleRtpRtcpImpl.
This ended up with needing to fork the current implementation
in order to not break downstream projects that were inheriting
from it. While those get updated, we'll move on with the forked
class.

Bug: webrtc:11581,b/8278269
Change-Id: I05b596cbda71aa5b72894c31a7119d17d4761883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175500
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31334}
2020-05-20 15:45:21 +00:00
1c7a6589a9 Add test for relay bandwidth capping.
Feature was added in
https://webrtc-review.googlesource.com/c/src/+/171226

Bug: webrtc:11434
Change-Id: Iee1e350976ab4043f15c5932cdc4f53b413bb302
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171861
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30940}
2020-03-30 13:02:46 +00:00
71fda3613c Extend NetworkRoute with more info about local/remote endpoints
This patch extends the NetworkRoute struct with more information
about local/remote endpoints. It adds
- adapter type
- adapter id
- relay

(previously it was "only" network_id)

The patch leaves the {local/remote}_network_id fields
around and populated since downstream projects depend
on them. They will be removed once they have migrated.

OWNER: srte@ call/ test/
OWNER: asapersson@ video/
OWNER: hta@ p2p/ pc/ rtc_base/

BUG: webrtc:11434
Change-Id: I9bcec385b40d707db385fef40b2c7a315dd35dd0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170628
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30848}
2020-03-20 16:55:38 +00:00
eac08bfe23 Reland "Wire up internal libvpx VP9 scaler to statistics proxy"
This reverts commit a2cb93d8b9659292f7ec73db53421d481f84c22c.

Reason for revert: Reland with no changes after downstream projects are
updated.

Original change's description:
> Revert "Wire up internal libvpx VP9 scaler to statistics proxy"
> 
> This reverts commit 50327a51007c3e25bc3bcd35b5d0945fe0f27d05.
> 
> Reason for revert: Breaks downstream tests
> 
> Original change's description:
> > Wire up internal libvpx VP9 scaler to statistics proxy
> > 
> > Bug: webrtc:11396
> > Change-Id: I5ac69208b00cc75d4e5dbb3ab86f234b3e1f29f8
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169922
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30725}
> 
> TBR=ilnik@webrtc.org,hbos@webrtc.org,nisse@webrtc.org
> 
> Change-Id: I53dcb41bdf8f8dccfcd43b717509ec047f590648
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:11396
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170102
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30734}

TBR=ilnik@webrtc.org,hbos@webrtc.org,nisse@webrtc.org,srte@webrtc.org

Change-Id: Ie47df4aec199701256c1dba8fa64176683becabc
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11396
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170105
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30738}
2020-03-10 11:15:51 +00:00
a2cb93d8b9 Revert "Wire up internal libvpx VP9 scaler to statistics proxy"
This reverts commit 50327a51007c3e25bc3bcd35b5d0945fe0f27d05.

Reason for revert: Breaks downstream tests

Original change's description:
> Wire up internal libvpx VP9 scaler to statistics proxy
> 
> Bug: webrtc:11396
> Change-Id: I5ac69208b00cc75d4e5dbb3ab86f234b3e1f29f8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169922
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30725}

TBR=ilnik@webrtc.org,hbos@webrtc.org,nisse@webrtc.org

Change-Id: I53dcb41bdf8f8dccfcd43b717509ec047f590648
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11396
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170102
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30734}
2020-03-10 08:09:50 +00:00
50327a5100 Wire up internal libvpx VP9 scaler to statistics proxy
Bug: webrtc:11396
Change-Id: I5ac69208b00cc75d4e5dbb3ab86f234b3e1f29f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169922
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30725}
2020-03-09 13:47:25 +00:00
40b764a6ba VideoSendStreamTest: remove unused array and member.
Bug: none
Change-Id: I9049be00ba461e5212406c9a5b51c67ba98240ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168947
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30624}
2020-02-27 08:38:51 +00:00
0e089db913 Roll chromium_revision ce459ab383..6d60176510 (742528:743892)
Manual changes:
  - Changed git repos for libcxx, libcxxabi and libunwind since they
  changed in Chromium.
  - Suppressed failing test on MSAN.

Change log: ce459ab383..6d60176510
Full diff: ce459ab383..6d60176510

Changed dependencies
* src/base: 1d6cd336dc..0794106942
* src/build: 188f078b2d..3e271e1ba5
* src/buildtools: afc5b798c7..feb2d0c562
* src/buildtools/linux64: git_revision:97cc440d84f050f99ff0161f9414bfa2ffa38f65..git_revision:4166e9fbc1fa5ceab69b69710a0f8b430c50127b
* src/buildtools/mac: git_revision:97cc440d84f050f99ff0161f9414bfa2ffa38f65..git_revision:4166e9fbc1fa5ceab69b69710a0f8b430c50127b
* src/buildtools/third_party/libc++/trunk: 78d6a7767e..d9040c75cf
* src/buildtools/third_party/libc++abi/trunk: 0d529660e3..196ba1aaa8
* src/buildtools/third_party/libunwind/trunk: 69d9b84cca..d999d54f4b
* src/buildtools/win: git_revision:97cc440d84f050f99ff0161f9414bfa2ffa38f65..git_revision:4166e9fbc1fa5ceab69b69710a0f8b430c50127b
* src/ios: 084a00adec..c5aa761a80
* src/testing: 688f493e49..f07276793c
* src/third_party: c6a4254b5e..f4d9303129
* src/third_party/android_deps/libs/com_google_dagger_dagger: version:2.17-cr0..version:2.26-cr0
* src/third_party/android_deps/libs/com_google_dagger_dagger_compiler: version:2.17-cr0..version:2.26-cr0
* src/third_party/android_deps/libs/com_google_dagger_dagger_producers: version:2.17-cr0..version:2.26-cr0
* src/third_party/android_deps/libs/com_google_dagger_dagger_spi: version:2.17-cr0..version:2.26-cr0
* src/third_party/android_deps/libs/com_google_guava_guava: version:27.0.1-jre-cr0..version:27.1-jre-cr0
* src/third_party/android_deps/libs/com_squareup_javapoet: version:1.11.0-cr0..version:1.11.1-cr0
* src/third_party/android_deps/libs/org_checkerframework_checker_compat_qual: version:2.3.0-cr0..version:2.5.3-cr0
* src/third_party/android_deps/libs/org_jetbrains_kotlin_kotlin_stdlib: version:1.3.41-cr0..version:1.3.50-cr0
* src/third_party/android_deps/libs/org_jetbrains_kotlin_kotlin_stdlib_common: version:1.3.41-cr0..version:1.3.50-cr0
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/7e43e2e8ee..6432bb46ab
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/9508452e18..d5a5c48017
* src/third_party/depot_tools: 10e0e6d6c1..1773f37de6
* src/third_party/ffmpeg: bcc5d9fec0..545152f302
* src/third_party/freetype/src: d09e831559..fa147af4a5
* src/third_party/libvpx/source/libvpx: 36133b04c0..55f2e4a0a8
* src/tools: af708e0676..e64334fd9c
Added dependencies
* src/third_party/android_deps/libs/org_jetbrains_kotlinx_kotlinx_metadata_jvm
* src/third_party/android_deps/libs/net_ltgt_gradle_incap_incap
DEPS diff: ce459ab383..6d60176510/DEPS

No update to Clang.

TBR=phoglund@webrtc.org,marpan@webrtc.org, jianj@chromium.org,
BUG=webrtc:11376

Change-Id: I5c45376e397c4ce6f9c151626b2280c750ca420c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168946
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30591}
2020-02-24 13:08:34 +00:00
5c35f2fb1b Delete RtpDepacketizerVp9 in favor of VideoRtpDepacketizerVp9
Bug: webrtc:11152
Change-Id: Ic50f2dc49ca420b3406d4dea11ed20328aa59136
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165382
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30195}
2020-01-09 13:07:44 +00:00
499de2d395 Fix tiny race condition when Vp9NonFlexMode_* tests were failing.
TSAN complained at each failure, which only added confusion.

Bug: webrtc:11243
Change-Id: Ic36ec13562699478be23050a7b723e1b9bb47f0e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164528
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#30161}
2020-01-07 09:29:14 +00:00
c33e4910c5 Prevent 'use after free' by waiting for all queued tasks to be processed.
Sometimes, a task bound to VideoSendStreamTest was called
after the underlying object had been destructed:
 1. |test| goes out of scope.
 2. There might still have been a task in fixture's queue,
    setup by OnSendRtp (capturing [this]) and invoked before
    the destruction of the fixture.

This CL uses the same workaround than BandwidthStatsTest:
block until all posted tasks are processed.

This fixes the following flaky tests:
 * VideoSendStreamTest.ChangingNetworkRoute
 * VideoSendStreamTest.RespectsMinTransmitBitrate*

Bug: webrtc:11156
Change-Id: I229c96d2abbbb60b43e9d9f62ad112507a21fe48
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163984
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#30152}
2020-01-07 04:18:51 +00:00
33f9d2b383 Migrate WebRTC on FrameGeneratorInterface and remove FrameGenerator class
Bug: webrtc:10138
Change-Id: If85290581a72f81cf60181de7a7134cc9db7716e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161327
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30033}
2019-12-07 00:54:26 +00:00
c347585927 Use RtpPacket instead of legacy RtpHeaderParser in video/ tests
Bug: None
Change-Id: Ia35daa58aae51becef40910187006398d825c5b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161331
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30026}
2019-12-06 10:54:39 +00:00
8ae70f6a30 Enable WebRTC-Bwe-MaxRttLimit by default.
Some of the field trial default values are changed as well.

Now available bitrate estimation will be decreasing when RTT is more than 3 seconds.
Unless different parameters for the field trial are specified.

Bug: None
Change-Id: Icd1923fc2e2e7766a7f645016c5432a52537145f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158840
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Konrad Hofbauer <hofbauer@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Nikita Zetilov <zetilovn@google.com>
Cr-Commit-Position: refs/heads/master@{#29823}
2019-11-18 16:53:11 +00:00
d15a0283d1 Hide deprecated SingleThreadedTaskQueueForTest behind an accessor
this change is intentionally noop.
Goal is to minimize change that would replace the
SingleThreadedTaskQueueForTest with a regular task queue.

Bug: webrtc:10933
Change-Id: I6da768941af048de3716af13e41b8f0f1ccd4cab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157892
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29569}
2019-10-22 11:57:49 +00:00
85a10001a5 Use deprecated SingleThreadedTaskQueueForTesting as regular task queue
Bug: webrtc:10933
Change-Id: I749ecd9cedb6798f1640ce663c6ebb6679889b67
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157883
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29565}
2019-10-22 08:34:57 +00:00
82a3f0ad7f Replace SingleThreadedTaskQueueForTesting::SendTask usage with ::webrtc::SendTask
Bug: webrtc:10933
Change-Id: I60738434b46e77b4644173ad168bc0efa58459b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156001
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29551}
2019-10-21 08:45:02 +00:00
c06aef2ad1 Reland "Use just a lookup map of RTP modules in PacketRouter"
This is a reland of 96f3de094566f32d842be6dd0906f1d13b8c8825
Downstream test is fixed, this is a pure reland.

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

Original change's description:
> Use just a lookup map of RTP modules in PacketRouter
>
> Since SSRCs of RTP modules are now set at construction time, we can
> use just a simple unordered map from SSRC to module in packet router.
>
> Bug: webrtc:11036
> Change-Id: I0b3527f17c9ee2df9253c778e5b9e3651a70b355
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155965
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29510}

Bug: webrtc:11036
Change-Id: I0731339dfd0781cc7f2f7ca78ac903539f25ff9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157304
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29514}
2019-10-17 12:59:39 +00:00
fbe84ef80f Revert "Use just a lookup map of RTP modules in PacketRouter"
This reverts commit 96f3de094566f32d842be6dd0906f1d13b8c8825.

Reason for revert: Downstream test is borked.

Original change's description:
> Use just a lookup map of RTP modules in PacketRouter
> 
> Since SSRCs of RTP modules are now set at construction time, we can
> use just a simple unordered map from SSRC to module in packet router.
> 
> Bug: webrtc:11036
> Change-Id: I0b3527f17c9ee2df9253c778e5b9e3651a70b355
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155965
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29510}

TBR=danilchap@webrtc.org,sprang@webrtc.org,srte@webrtc.org

Change-Id: I31330fd68ab809ff3951573791e9a79b81599958
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11036
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157281
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29511}
2019-10-17 11:17:41 +00:00
96f3de0945 Use just a lookup map of RTP modules in PacketRouter
Since SSRCs of RTP modules are now set at construction time, we can
use just a simple unordered map from SSRC to module in packet router.

Bug: webrtc:11036
Change-Id: I0b3527f17c9ee2df9253c778e5b9e3651a70b355
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155965
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29510}
2019-10-17 11:06:34 +00:00
6841d25d45 Reland "RtpRtcp modules and below: Make media, RTX and FEC SSRCs const"
This is a reland of 17608dc4592fe25c1effdd75bf856f4af251942e

Downstream test now fixed.
As a precaution, also avoid DCHECKS for non-zero SSRC.
First patch set is reland, second makes checks more lenient.

Original change's description:
> RtpRtcp modules and below: Make media, RTX and FEC SSRCs const
>
> Downstream usage of SetSsrc() / SetRtxSsrc() should now be gone. Let's
> remove them, make the members const, and remove now unnecessary locking.
>
> Bug: webrtc:10774
> Change-Id: Ie4c1b3935508cf329c5553030f740c565d32e04b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155660
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29475}

Bug: webrtc:10774
Change-Id: I540b49a31a31e98d87f02ae04083d5206e71c1b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157100
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29491}
2019-10-15 14:03:19 +00:00
e8a6bc3f25 Revert "Reland "RtpRtcp modules and below: Make media, RTX and FEC SSRCs const""
This reverts commit c9348218cfe0cff6d0d3a383f7d1d6cfce4b1262.

Reason for revert: Downstream tests are relying on incorrect behavior which this CL explicitly checks...

Original change's description:
> Reland "RtpRtcp modules and below: Make media, RTX and FEC SSRCs const"
> 
> This is a reland of 17608dc4592fe25c1effdd75bf856f4af251942e
> 
> Downstream fixed, relanding.
> 
> Original change's description:
> > RtpRtcp modules and below: Make media, RTX and FEC SSRCs const
> >
> > Downstream usage of SetSsrc() / SetRtxSsrc() should now be gone. Let's
> > remove them, make the members const, and remove now unnecessary locking.
> >
> > Bug: webrtc:10774
> > Change-Id: Ie4c1b3935508cf329c5553030f740c565d32e04b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155660
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#29475}
> 
> TBR=nisse@webrtc.org
> 
> Bug: webrtc:10774
> Change-Id: I759bed3ff1909857696c6d1b13df595a5e552f03
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157049
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29486}

TBR=nisse@webrtc.org,sprang@webrtc.org

Change-Id: I168fb3738a04dfdbd1581ddd8c3276ede9f72322
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10774
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157080
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29488}
2019-10-15 11:54:33 +00:00
c9348218cf Reland "RtpRtcp modules and below: Make media, RTX and FEC SSRCs const"
This is a reland of 17608dc4592fe25c1effdd75bf856f4af251942e

Downstream fixed, relanding.

Original change's description:
> RtpRtcp modules and below: Make media, RTX and FEC SSRCs const
>
> Downstream usage of SetSsrc() / SetRtxSsrc() should now be gone. Let's
> remove them, make the members const, and remove now unnecessary locking.
>
> Bug: webrtc:10774
> Change-Id: Ie4c1b3935508cf329c5553030f740c565d32e04b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155660
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29475}

TBR=nisse@webrtc.org

Bug: webrtc:10774
Change-Id: I759bed3ff1909857696c6d1b13df595a5e552f03
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157049
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29486}
2019-10-15 11:42:05 +00:00
4ed0b52c12 Revert "RtpRtcp modules and below: Make media, RTX and FEC SSRCs const"
This reverts commit 17608dc4592fe25c1effdd75bf856f4af251942e.

Reason for revert: Breaks downstream build

Original change's description:
> RtpRtcp modules and below: Make media, RTX and FEC SSRCs const
> 
> Downstream usage of SetSsrc() / SetRtxSsrc() should now be gone. Let's
> remove them, make the members const, and remove now unnecessary locking.
> 
> Bug: webrtc:10774
> Change-Id: Ie4c1b3935508cf329c5553030f740c565d32e04b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155660
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29475}

TBR=nisse@webrtc.org,sprang@webrtc.org

Change-Id: Idc60f26f34dd0456a40c72375ae829e25b28621f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10774
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157046
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29483}
2019-10-15 09:43:21 +00:00
17608dc459 RtpRtcp modules and below: Make media, RTX and FEC SSRCs const
Downstream usage of SetSsrc() / SetRtxSsrc() should now be gone. Let's
remove them, make the members const, and remove now unnecessary locking.

Bug: webrtc:10774
Change-Id: Ie4c1b3935508cf329c5553030f740c565d32e04b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155660
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29475}
2019-10-15 07:50:59 +00:00
24d251f796 Add 100 ms network delay to the SupportsFlexFEC* tests.
Some of the tests are currently flaky because FEC is disabled if the
RTT is <200 ms, and the simulated network is configured to use 100 ms
for the send transport, but nothing is configured for the receive
transport. This CL configures the receive transport to 100 ms delay.

Bug: webrtc:10920
Change-Id: I79995693ba73683406fa9ced92a7918e6c05473f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154571
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29394}
2019-10-07 13:01:05 +00:00
44db436e87 Propagate task queue to create test::DirectTransport by TaskQueueBase interface
actual task queue implementation for these tests is intentionally unchanged for now.

while at it, change return type of created transports to unique_ptr to note passing ownership.

Bug: webrtc:10933
Change-Id: I324597b503e647c471f43511340eb9c07ba03ee8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154743
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29335}
2019-09-30 03:23:07 +00:00
317a1f09ed Use std::make_unique instead of absl::make_unique.
WebRTC is now using C++14 so there is no need to use the Abseil version
of std::make_unique.

This CL has been created with the following steps:

git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt
git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt
git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt

diff --new-line-format="" --unchanged-line-format="" \
  /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \
  uniq > /tmp/only_make_unique.txt
diff --new-line-format="" --unchanged-line-format="" \
  /tmp/only_make_unique.txt /tmp/memory.txt | \
  xargs grep -l "absl/memory" > /tmp/add-memory.txt

git grep -l "\babsl::make_unique\b" | \
  xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g"

git checkout PRESUBMIT.py abseil-in-webrtc.md

cat /tmp/add-memory.txt | \
  xargs sed -i \
  's/#include "absl\/memory\/memory.h"/#include <memory>/g'
git cl format
# Manual fix order of the new inserted #include <memory>

cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \
  xargs sed -i '/#include "absl\/memory\/memory.h"/d'

git ls-files | grep BUILD.gn | \
  xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d'

python tools_webrtc/gn_check_autofix.py \
  -m tryserver.webrtc -b linux_rel

# Repead the gn_check_autofix step for other platforms

git ls-files | grep BUILD.gn | \
  xargs sed -i 's/absl\/memory:memory/absl\/memory/g'
git cl format

Bug: webrtc:10945
Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 15:47:29 +00:00
6516f76f9b Deprecate SingleThreadedTaskQueueForTesting class.
This class doesn't strictly follow rtc::TaskQueue semantics,
which makes it surprising and hard to use correctly.
Please use TaskQueueForTest instead.

This CL follows usual deprecation process:

1/ Rename.
% for i in `git ls-files` ; sed -i "s:SingleThreadedTaskQueueForTesting:DEPRECATED_SingleThreadedTaskQueueForTesting:" $i

2/ Annotate old name for downstream users and accidental new uses.

Bug: webrtc:10933
Change-Id: I80b4ee5a48df1f63f63a43ed0efdb50eb7fb156a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150788
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#29045}
2019-09-03 10:31:30 +00:00
25eb47ccf1 Make the RtpHeaderParserImpl available to tests and tools only.
There are a few reasons for making this test only:
* The code is only used by tests and utilities.
* The pure interface has only a single implementation so an interface isn't really needed.
  (a followup change could remove it altogether)
* The implementation always incorporates locking regardless of how the class gets used.
  See e.g. previous use in the Packet class.
* The implementation is a layer on top of RtpUtility::RtpHeaderParser which is
  sufficient for most production cases.

Change-Id: Ide6d50567cf8ae5127a2eb04cceeb10cf317ec36
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150658
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29010}
2019-08-29 15:56:40 +00:00