Commit graph

34399 commits

Author SHA1 Message Date
Christoffer Jansson
2bcdb5d63a Remove phoglund from ENG_REVIEW_OWNERS
Bug: NONE
Change-Id: Iea7196f9d163a4efcd8d08799c14fb96581197ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232300
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#35167}
2021-10-08 08:29:42 +00:00
webrtc-version-updater
dd410c007b Update WebRTC code version (2021-10-08T04:04:59).
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org

Bug: None
Change-Id: Ib23a61b5d45059912e8076f042f89bc3f98858f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234560
Reviewed-by: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#35166}
2021-10-08 05:24:53 +00:00
Byoungchan Lee
7284bd4dab Use GCD instead of Detached Thread in Async Resolver when on MacOS/iOS
The advantage is that GCD maintains the internal thread pool and
spawns threads when needed. I would expect the behavior to be
almost identical to creating a thread using PlatformThread.

Bug: webrtc:13237
Change-Id: Ie4406b5d128c244f66a73830d5a27f2d8fd88549
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234300
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#35165}
2021-10-08 02:47:51 +00:00
Christoffer Rodbro
40abb7d8ff Default the behavior allowing fast rampup when REMB cap is lifted.
Bug: none
Change-Id: I60d5ed448b3cfb6591bd77b97f406a62e2fdd704
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234523
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35164}
2021-10-07 14:44:13 +00:00
Alessio Bazzica
5c3ae49b44 AudioFrameView: size_t -> int
Bug: webrtc:7494
Change-Id: I46b1328f3d7da721e144cc3752ed4f458084cf62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234522
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35163}
2021-10-07 14:41:03 +00:00
Alessio Bazzica
82ea4ee9bf AGC2 refactoring: better names for GainController2 members
Bug: webrtc:7494
Change-Id: Ibac8a3953e68fa7bdbddfb9d4eb24f2712ba05b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234480
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35162}
2021-10-07 13:17:35 +00:00
Sergey Silkin
9b2a7461f0 Use fallback encoder if primary can't be created
In case if primary encoder can't be instantiated (max number of
instances has reached, for example), use fallback encoder.

Bug: none
Change-Id: I477bdeb7af4dcce50e36b1804ffc6ad2ab004dfd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234500
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35161}
2021-10-07 11:42:26 +00:00
webrtc-version-updater
bcef6e1859 Update WebRTC code version (2021-10-07T04:02:51).
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org

Bug: None
Change-Id: Icdd28fc59b8ba58521637576956902506032cb60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234460
Reviewed-by: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#35160}
2021-10-07 06:02:57 +00:00
Hanna Silen
5c7d5c9ce1 AudioProcessingImpl: Move analog gain change check
Move the check for analog gain changes so that it can be used
independently of echo_controller. This change is needed to land
https://webrtc-review.googlesource.com/c/src/+/234140.

Bug: webrtc:12774
Change-Id: I9ea127b0a4d374f31493d6f8afcacee40fa9257c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234383
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35159}
2021-10-06 17:13:52 +00:00
Nico Weber
bde89ab09e win: Disable deprecation warning for one call of GetVersionEx
Like https://chromium-review.googlesource.com/c/chromium/src/+/3207949
but using pragmas that work with both cl.exe and (very new versions of)
clang-cl.

webrtc also needs the granularity, see e.g.
https://webrtc-review.googlesource.com/c/src/+/229140

Bug: chromium:1255114
Change-Id: I6a2bf9447f377988a2a3844d6ef16aeee63734f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234420
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35158}
2021-10-06 16:32:52 +00:00
Björn Terelius
fc5a4f74ac Revert "Use AsyncDnsResolver API in bindings and tests"
This reverts commit a0577605b0.

Reason for revert: Speculative revert due to downstream tests

Original change's description:
> Use AsyncDnsResolver API in bindings and tests
>
> Bug: webrtc:12598
> Change-Id: Ia4db91bf6dcd257cd85f4089dee4c7bbea433216
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234342
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Reviewed-by: Markus Handell <handellm@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35154}

TBR=hta@webrtc.org,handellm@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: I57f3ff70b6374e7be670526a90dfb1651e9b1148
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12598
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234382
Reviewed-by: Björn Terelius <terelius@google.com>
Commit-Queue: Björn Terelius <terelius@google.com>
Cr-Commit-Position: refs/heads/main@{#35157}
2021-10-06 15:05:57 +00:00
Jan Grulich
3695640504 PipeWire capturer: copy content from PW buffer directly to DesktopFrame
This avoids an additional step where we originally copied content from
PipeWire buffer to a temporary location and from there to DesktopFrame.
This results into less copy operations and hopefully to faster
screensharing.

I didn't do some exact measures, but simply running htop while sharing a
4k screen I can see following results (usage per top 5 processes):
1) Without this change - 66%, 64%, 26% 23%, 10%
2) With this change - 41%, 39%, 19%, 17%, 12%,

Bug: webrtc:13239
Change-Id: I6a661ecc96bfeef370c1a5a3b9dc5e3c0fc665c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231684
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35156}
2021-10-06 13:22:32 +00:00
Niels Möller
6d19d14c26 Add AsyncListenSocket, as alias for AsyncPacketSocket
A preparation for splitting server sockets out into a separate
interface, see https://webrtc-review.googlesource.com/c/src/+/232607.

Transition plan:

1. Land this cl.

2. Update downstream code to use the new name.

3. Attempt landing
   https://webrtc-review.googlesource.com/c/src/+/232607. May need
   additional steps to not break downstream implementations of
   PacketSocketFactory::CreateServerTcpSocket.

Bug: webrtc:13065
Change-Id: Ife448c705222f4c9f66a096e3dc7eb07e0f9c3af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233700
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35155}
2021-10-06 11:42:50 +00:00
Harald Alvestrand
a0577605b0 Use AsyncDnsResolver API in bindings and tests
Bug: webrtc:12598
Change-Id: Ia4db91bf6dcd257cd85f4089dee4c7bbea433216
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234342
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35154}
2021-10-06 11:18:30 +00:00
Robert Mader
408e4da26f Pipewire: Do not typecheck the portal session_handle
Desktop sharing via Pipewire will break for clients updating to
xdg-desktop-portal 1.10 due to a bug fix in the API implementation[1].

This ports over a fix from OBS Studio[2] that also is used in the
downstream Firefox WebRTC copy[3].

1: https://github.com/flatpak/xdg-desktop-portal/pull/609
2: https://github.com/obsproject/obs-studio/pull/5294
3: https://phabricator.services.mozilla.com/D126053
Bug: webrtc:13192
Change-Id: I497dd1bb53cc39dee3732c2e0014e2e36a7afb6c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232329
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35153}
2021-10-06 10:32:55 +00:00
Jan Grulich
e1e05afec7 Reland "PipeWire capturer: implement proper DMA-BUFs support""
This is a reland of f2177f6612

Original change's description:
> PipeWire capturer: implement proper DMA-BUFs support
>
> Currently both KWin (KDE) and Mutter (GNOME) window managers don't
> use DMA-BUFs by default, but only when client asks specifically for
> them (KWin) or when experimental DMA-BUF support is enabled (Mutter).
> While current implementation works just fine on integrated graphics
> cards, it causes issues on dedicated GPUs (AMD and NVidia) where the
> code either crashes or screensharing is slow and unusable.
>
> To fix this, DMA-BUFs has to be opened using OpenGL context and not
> being directly mmaped(). This implementation requires to use DMA-BUF
> modifiers, as they are now mandatory for DMA-BUFs usage.
>
> Documentation for this behavior can be found here:
> https://gitlab.freedesktop.org/pipewire/pipewire/-/blob/master/doc/dma-buf.dox
>
> Bug: chromium:1233417
> Change-Id: I0cecf16d6bb0f576954b9e8f071cab526f7baf2c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227022
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Tommi <tommi@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34889}

Bug: chromium:1233417
Change-Id: I308501d86ec18ab6df9bcee569c4b72df7926549
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231180
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35152}
2021-10-06 10:31:51 +00:00
Niels Möller
404cd60ecc Fix weird socket member naming in AsyncStunTCPSocketTest
Bug: webrtc:13065
Change-Id: Ifd7af4b283d55cbe0e3a03185b1b8e0bab6d47cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234322
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35151}
2021-10-06 08:52:51 +00:00
Erik Språng
04c911c6a5 Revert "Turn on WebRTC-TaskQueuePacer by defualt."
This reverts commit b251145e38.

Reason for revert: Downstream issue

Original change's description:
> Turn on WebRTC-TaskQueuePacer by defualt.
>
> Bug: webrtc:10809
> Change-Id: If58ae3d9debc69ee68e6aeb6cecf010e60f6426f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233580
> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35145}

TBR=sprang@webrtc.org,crodbro@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: I3a8db6fc2376ccc528f1e2fa6acc08ce05afebbf
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10809
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234340
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35150}
2021-10-06 08:15:23 +00:00
Minyue Li
2bfa5b20fe Default sending capture clock offset in abs-capture-time header extension.
Bug: webrtc:10739
Change-Id: Ieadb6d75122e5988b22509ac14dc528277a7f56f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232906
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35149}
2021-10-06 07:53:32 +00:00
Harald Alvestrand
c9f43f8f81 Use AsyncDnsResolver in TurnPort class
Bug: webrtc:12598
Change-Id: Ie53c27d3a614521f4a8b665fd321b1db53dc70b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234261
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35148}
2021-10-06 07:28:26 +00:00
Harald Alvestrand
b7b306bab5 Use AsyncDnsResolver in UDPPort class
Bug: webrtc:12598
Change-Id: I408d7daa0f3b5df6f45bcc97fa445bc8158b54ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233561
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35147}
2021-10-06 05:15:11 +00:00
Nico Weber
79bd4f1bc3 win: Consolidate on a single version checking API
No intended behavior change.

Happens to remove one call to GetVersionEx.

Bug: chromium:1255114
Change-Id: If4d1c57fa27ad4a7547f8f18c3abe38bc9b2a325
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234160
Reviewed-by: Joe Downing <joedow@chromium.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35146}
2021-10-05 22:13:37 +00:00
Erik Språng
b251145e38 Turn on WebRTC-TaskQueuePacer by defualt.
Bug: webrtc:10809
Change-Id: If58ae3d9debc69ee68e6aeb6cecf010e60f6426f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233580
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35145}
2021-10-05 16:17:09 +00:00
Alessio Bazzica
7b80d4480e AGC2: SIMD allowed config flags to field trials
Bug: webrtc:7494
Change-Id: I41fa05d2ef6d969750f3d4c1e40ecbcd30293b5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233741
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35144}
2021-10-05 12:01:38 +00:00
Erik Språng
41bbc3df78 Fix bug in dynamic pacer causing slightly inaccurate pacing rate.
When new packets are enqueued after a dead-period where media debt is
zero, that time slice should not be used to reduce the debt for the
new packet.

Bug: webrtc:10809
Change-Id: Ifb960548e6aa349b79f37743cbfed78a5c937a13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234081
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35143}
2021-10-05 10:56:02 +00:00
Alessio Bazzica
b8ffdc4bb3 APM: fix CaptureLevelAdjustment::operator==
Bug: webrtc:7494
Change-Id: I0ea13af2e23ed1490fa22d75d104bdd45b0452bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233460
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35142}
2021-10-05 07:25:06 +00:00
webrtc-version-updater
958772efc5 Update WebRTC code version (2021-10-05T04:03:48).
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org

Bug: None
Change-Id: I9b83673348780f33151354ff20fa2ee492151870
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234048
Reviewed-by: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#35141}
2021-10-05 05:27:05 +00:00
Alessio Bazzica
a850e6c8b6 AGC2 config: allow tuning of headroom, max gain and initial gain
This CL does *not* change the behavior of the AGC2 adaptive digital
controller - bitexactness verified with audioproc_f on a collection of
AEC dumps and Wav files (42 recordings in total).

Tested: compiled Chrome with this patch and made an appr.tc test call

Bug: webrtc:7494
Change-Id: Ia8a9f6fbc3a3459b888a2eed87e108f0d39cfe99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233520
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35140}
2021-10-04 16:11:00 +00:00
Vojin Ilic
41b4397e1a Use accumulate to calculate recv_delta_size
It's a modern way to sum element of an a array.

Bug: None
Change-Id: Idb09442b4647b4be9771f64a7a561b305bd9aa6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233942
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35139}
2021-10-04 12:11:41 +00:00
Asa Persson
606d3cb1cf VideoStreamEncoderTest: Use DataRate for some constants.
Use config from FakeEncoder in some tests.

Bug: none
Change-Id: I1d7e01f604f8aabb5d6815bb519ef2532d024d76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233243
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35138}
2021-10-04 10:10:40 +00:00
webrtc-version-updater
c89560146b Update WebRTC code version (2021-10-04T04:04:13).
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org

Bug: None
Change-Id: I7145dea736cf03a1cfe032b5df00342ad7073408
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233895
Reviewed-by: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#35137}
2021-10-04 05:33:48 +00:00
Keiichi Enomoto
75d0de320f Roll src/third_party/libjpeg_turbo/ ff19e5b2e..49836d72b (1 commit)
ff19e5b2e1..49836d72bd

$ git log ff19e5b2e..49836d72b --date=short --no-merges --format='%ad %ae %s'
2021-10-01 enm10k Add MANGLE_JPEG_NAMES to public_configs.

Created with:
  roll-dep src/third_party/libjpeg_turbo

Bug: webrtc:13101
Change-Id: I803d2df02ab2050cf37baee545ce936152962686
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233800
Reviewed-by: Mirko Bonadei <mbonadei@google.com>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35136}
2021-10-03 16:39:06 +00:00
Mirko Bonadei
cc99299bbc Remove use_xcode_clang=true from iOS packaging script.
Bug: webrtc:13197
Change-Id: I90a71cf1a1af9ba372cf9d23b73b9aeb3ea7b0b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232600
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35135}
2021-10-03 12:19:46 +00:00
Mirko Bonadei
54c90f2330 [-Wshadow] - Fix some warnings.
First CL to try to understand the extent of the cleanup needed in
order to remove -Wno-shadow and follow Chromium on enabling this
diagnostic.

Bug: webrtc:13219
Change-Id: Ie699762da50fe3dbc08b1fd92220962d4b7da86b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233641
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35134}
2021-10-03 11:53:16 +00:00
webrtc-version-updater
e3d26f534d Update WebRTC code version (2021-10-03T04:05:03).
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org

Bug: None
Change-Id: Ibcf533d4b73ed01ef503a19b52e7bc22786fceeb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233885
Reviewed-by: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#35133}
2021-10-03 05:45:08 +00:00
Ilya Nikolaevskiy
db94869ca3 Make CroppedWindowCapturer more resilient
Bug: chromium:1245272
Change-Id: I276c98ad0aea3dd0e614b935b9a7566c77d5026a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233720
Reviewed-by: Joe Downing <joedow@chromium.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35132}
2021-10-01 18:23:59 +00:00
Harald Alvestrand
985310ea3b Add CreateAsyncDnsResolver to PacketSocketFactory API
This unlocks migration from AsyncResolver to AsyncDnsResolver for
clients that implement PacketSocketFactory.

A default implementation is provided, so that clients that implement
CreateAsyncResolver will still see their name resolution work.

Bug: webrtc:12598
Change-Id: If835cbc753712e9f5b4bd3d5805c7f7d2a561ee5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233500
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35131}
2021-10-01 16:12:50 +00:00
Philipp Hancke
ae566cd831 audio/red: provide default fmtp line
otherwise the generated codec won't match the preassigned codec
and red will use 96 as payload type, increasing the payload type
congestion in the upper range.

BUG=webrtc:11640

Change-Id: I466ed6d4e025ef116f3099e85855e10493408ab1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233560
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35130}
2021-10-01 13:02:30 +00:00
Björn Terelius
1d4aa36988 Allow encoding string fields in new event log format.
Return parse results as a StatusOr containing views to values owned by the parser.

Bug: webrtc:11933
Change-Id: Icf26b9cb651d1e9244c764c3ec1fdb66abfc9e08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233740
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35129}
2021-10-01 12:26:10 +00:00
webrtc-version-updater
ba4d870acf Update WebRTC code version (2021-10-01T04:03:55).
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org

Bug: None
Change-Id: Ibdac381e779ec104a02465815df7c123f8d99760
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233606
Reviewed-by: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#35128}
2021-10-01 05:34:50 +00:00
Victor Boivie
5755f3edaf dcsctp: Add sequence checker to socket
The DcSctpSocket is not thread safe and must be called from a single
thread or from a task queue that serializes access to it. This is now
validated at run-time in debug builds.

Bug: None
Change-Id: I3ed816924c20f6ed7e84a3273bee5a3f8f74112b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233420
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35127}
2021-09-30 11:25:18 +00:00
Björn Terelius
fe903d5eab Add encoding for numeric RTC event fields.
Bug: webrtc:11933
Change-Id: I5fe98c6753547b2c096d8e97870a7f9ce90b7b8b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230703
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35126}
2021-09-30 11:04:59 +00:00
Niels Möller
a654e07d11 Eliminate a temporary std::string in ParsedFailed helper.
Bug: None
Change-Id: If3435b5e9da9d2049c9b82b8b68e54d1ecc69003
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233440
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35125}
2021-09-30 09:31:28 +00:00
Victor Boivie
82ccdd36aa dcsctp: Add network/throughput tests
This initial version contains a few tests, testing both lossy, non-lossy
and bandwidth limited networks. It uses simulated time, and runs much
faster than wall time on release builds, but slower on debug when there
is a lot of outstanding data (high throughput) as there are consistency
checks on outstanding data. Because of that, some tests had to be
disabled in debug build.

Bug: webrtc:12943
Change-Id: I9323f2dc99bca4e40558d05a283e99ce7dded7f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225201
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35124}
2021-09-30 07:59:58 +00:00
Philipp Hancke
ee4c930b4e ice: fix comment about relay preference
BUG=webrtc:13195

Change-Id: I86b0ff259cec7d2dc95ba5d7ae1022aca4dcb01c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233260
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35123}
2021-09-30 07:45:19 +00:00
Niels Möller
d4aa3a3196 Use absl::string_view in SDP-related utilities
A step towards reduced copying in SDP parsing.

Bug: None
Change-Id: I3a5d89f483c1809929b7160b563c67b040c9df41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233080
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35122}
2021-09-30 07:44:09 +00:00
webrtc-version-updater
8afd22f286 Update WebRTC code version (2021-09-30T04:03:50).
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org

Bug: None
Change-Id: I2167226951b7b3f858052847066785091f1c20ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233386
Reviewed-by: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#35121}
2021-09-30 05:59:18 +00:00
Shuhai Peng
f270770679 video: Implement bandwidth based scaler
The |slice_qp_detla| reported by the hardware is not credible, which
causing the quality scaler cannot work properly,the resolution cannot
be adjusted correctly.

To fix this issue, this CL implements a bandwidth scaler which is used
for adjust resolution, this scaler will be used when QP based quality
scaler is not working due to untrusted QP reported by HW AVC encoder.

Bug: webrtc:12942
Change-Id: I2fc5f07a5400ec7e5ead2c2c502faee84d7f2a76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228860
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35120}
2021-09-29 10:39:27 +00:00
Johannes Kron
23bfff3383 Change default parameters for the low-latency video pipeline
min_pacing:8ms, to avoid the situation where bursts of frames are sent
to the decoder at once due to network jitter. The bursts of frames
caused the queues further down in the processing to be full and
therefore drop all frames.

max_decode_queue_size:8, in the event that too many frames have piled
up, do as before and send all frames to the decoder to avoid building
up any latency.

These setting only affect the low-latency video pipeline that is enabled
by setting the playout RTP header extension to min=0ms, max>0ms.

Bug: chromium:1138888
Change-Id: I8154bf3efe7450b770da8387f8fb6b23f6be26bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233220
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35119}
2021-09-29 09:53:17 +00:00
Niels Möller
aa373166f7 Pass a SocketFactory to BasicNetworkManager constructor
Used by QueryDefaultLocalAddress, instead of relying on the update
thread's associated socket server.

This is not the only use of rtc::Thread::socketserver() in the
BasicNetworkManager class. It also interacts with the thread's
socket server to call set_network_binder. That is unchanged by this cl,
perhaps those calls can be moved to the caller of StartNetworkMonitor and
StopNetworkMonitor.

Bug: webrtc:13145
Change-Id: If109c2dcb0e74b183e10bb3db7a5aefcc95d1a8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232613
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35118}
2021-09-29 08:59:37 +00:00