Commit Graph

35279 Commits

Author SHA1 Message Date
09f5713a6b Reland "Move VideoTrackSourceProxy creation into VideoTrack."
This is a reland of 4bc7223cf0775737a615c677b82c78e49a6a8a2c

Original change's description:
> Move VideoTrackSourceProxy creation into VideoTrack.
>
> This CL contains a part of a previously reviewed, landed, reverted,
> relanded and re-reverted CL:
> https://webrtc-review.googlesource.com/c/src/+/250180
>
> Bug: webrtc:13540
> Change-Id: Id6df8da5ff61e3ec90d0b3f3d828e8f670d4931b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251860
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36025}

Bug: webrtc:13540
Change-Id: I81df66daa40c59ab0b8677d7065f1703f071a10a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252041
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36047}
2022-02-22 11:14:15 +00:00
f1e4a66e94 Fix py3 compatibility for webrtc_version_updater
Bug: webrtc:13607
Change-Id: I5890dc86286b0a6ece793655bcda15315d415b39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252060
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#36046}
2022-02-22 10:59:46 +00:00
9bbfe9e5e0 dcsctp: Fix data race in debug logging
The variable instance_count might be accessed from multiple threads when
different PeerConnectionFactory objects are used, which may create
multiple network threads. This is a pattern mostly noticed in tests.

This fixes issues with logging when run under TSAN, it should not have
any production impact.

Bug: chromium:1243702
Change-Id: Iab1412a7907545811a309cab27a3ae23b4718606
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251983
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Victor Boivie <boivie@google.com>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36045}
2022-02-22 10:21:26 +00:00
7b0a30ec9a Allow low_bandwith_audio_test.py to pass unknown arg to the test.
* The idea is copied from flags_compatibility since this file does a bit the same thing.
* Remove extra_test_args which is not used and becomes unecessary.
* Fix lint issues.

Bug: b/197492097
Change-Id: I378e163a5116ded13619f91ce50859519c9550df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252004
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#36044}
2022-02-22 09:31:47 +00:00
c8f0502f0f Fix typo - Parenthesis was in the wrong place.
No-Try: True
Bug: webrtc:13607
Change-Id: Iba13bcf94f7a89cdc0e124d42e879b9c900a56cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252003
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#36043}
2022-02-22 08:09:14 +00:00
7a5978e4cd Revert "Add the possibility to output a json gtest output to the perf tests."
This reverts commit cbfa235b3548ae1a0b8f5b80a879214ca5f1bd56.

Reason for revert: iOS bots will use flags_compatibility with isolated_script_test_output but not gtest_output.

Original change's description:
> Add the possibility to output a json gtest output to the perf tests.
>
> We use the Chromium existing flag isolated_script_test_output that we translate into gtest_output.
> This is because the Chromium flag has the same purpose as gtest_output and is already provided in the recipe modules.
>
> No-Presubmit: True
> Bug: b/197492097
> Change-Id: Ia432a85b0e0ab32008b39ffe751d11aefb9b24ea
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251041
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Jeremy Leconte <jleconte@google.com>
> Cr-Commit-Position: refs/heads/main@{#35937}

No-Presubmit: True
Bug: b/197492097
Change-Id: I94e75328570f89011fbb0daf035f0072b8ea2f7d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252040
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#36042}
2022-02-22 07:57:34 +00:00
694eae7064 Decode/encode data to utf-8
Currently blocking the chromium to webrtc roller

Bug: webrtc:13607
Change-Id: I02237b266d4f2a2fdec2975b9e67d1c4b2099a48
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252001
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#36041}
2022-02-21 19:53:34 +00:00
3147e29c4e Refactor encoder-complexity param in VideoCodec w/backward compatibility
Move complexity parameter to the main VideoCodec class to enable
additional video codecs to use the parameter without creating a new
codec-specific structure.

Bug: webrtc:13694
Change-Id: Icb7cf640b178875d799f39ade8b5084e3222bb1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251921
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Michael Horowitz <mhoro@google.com>
Cr-Commit-Position: refs/heads/main@{#36040}
2022-02-21 19:40:44 +00:00
1f6e4308ab Revert "Move VideoTrackSourceProxy creation into VideoTrack."
This reverts commit 4bc7223cf0775737a615c677b82c78e49a6a8a2c.

Reason for revert: Regressions in PC tests https://crbug.com/webrtc/13697

Original change's description:
> Move VideoTrackSourceProxy creation into VideoTrack.
>
> This CL contains a part of a previously reviewed, landed, reverted,
> relanded and re-reverted CL:
> https://webrtc-review.googlesource.com/c/src/+/250180
>
> Bug: webrtc:13540
> Change-Id: Id6df8da5ff61e3ec90d0b3f3d828e8f670d4931b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251860
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36025}

Bug: webrtc:13540
Change-Id: Ibae8c1d39fa4d71aafd9666b66e6abc8eb8fddb2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251980
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36039}
2022-02-21 18:22:24 +00:00
c13caac2d5 Fix urllib import + add test
Unblocks the chromium to webrtc roller.

Bug: webrtc:13607
Change-Id: I877d8d25624bdc924425ef338f392ce7686207f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251984
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#36038}
2022-02-21 16:24:34 +00:00
808531653e In RtcpTransceiver implement handling incoming RRTR
Bug: webrtc:8239
Change-Id: I4a469b6a0c2e387e35262798f4686fbf310d00cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251902
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36037}
2022-02-21 14:07:34 +00:00
c4ed5f0b1a Adding fuzzer for G711/PCM u/A decoders and fixing a fuzzer problem
Bug: chromium:1279775
Change-Id: I8cc3f5fe25b9e707e9d171251026bd5a8bad5da5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251844
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36036}
2022-02-21 13:35:24 +00:00
153c9e5107 Revert "Use internal() in VideoTrack when invoking the source."
This reverts commit 962bf1896185c0d84232f2bfae492eeb04e1236d.

Reason for revert: Regressions in PC tests https://crbug.com/webrtc/13697

Original change's description:
> Use internal() in VideoTrack when invoking the source.
>
> This skips going through the proxy and potentially hide a thread hop
> should a regression occur.
>
> This CL contains a part of a previously reviewed, landed, reverted,
> relanded and re-reverted CL:
> https://webrtc-review.googlesource.com/c/src/+/250180
>
> Bug: webrtc:13540
> Change-Id: If098f5c04a263547fb53f44e9f9738b8e941a294
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251861
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36026}

Bug: webrtc:13540
Change-Id: Iea76094aedda91271154f89c356b140c95717976
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251981
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36035}
2022-02-21 13:31:24 +00:00
ba2677061a Add fuzzer test for G722 and fix a fuzzer problem
The problem was fixed by implementing the methid PacketDuration() in
AudioDecoderG722StereoImpl, which catches the issue in
AudioDecoder::Decode().


Bug: chromium:1280851
Change-Id: I31f974b9999f3c1c62b0e5dc39bb3e56a9a9388d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251842
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36034}
2022-02-21 10:16:47 +00:00
0b06552ab3 Android: Respect input buffer layout of MediaFormat
On Android, MediaCodec can request a specific layout of the input buffer.
One can use the stride and slice height to calculate the layout from
the Encoder's MediaFormat. The current code assumes
a specific layout, which is a problematic in Android 12.
Fix this by honoring the stride and slice-height.

Bug: webrtc:13427
Change-Id: I2d3e429309e3add3ae668e0390460b51e6a49eb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/240680
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#36033}
2022-02-21 09:52:54 +00:00
39f027e0b0 Add .mailmap for git.
This is purely to aid with `git log` type statements that allows for
grouping different display names for the same address.

No-try: true
Bug: none
Change-Id: I6b0af50eac356aa864e1387f3f35c3270c211faf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251941
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36032}
2022-02-20 14:22:13 +00:00
feac97bb25 AgcManagerDirect: improve AgcMinMicLevelExperimentEnabled50 test
Also test the field trial with valid parameter and non-empty suffix.

Bug: webrtc:7494
Change-Id: I3d871b41dd71c951ac56e180b3c09cda4c3627d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251441
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36031}
2022-02-18 17:52:01 +00:00
8968bcae8d In RtcpTransceiver avoid generating rtcp sender reports for inactive senders
Bug: webrtc:8239
Change-Id: I97d50c628db04c56669179ab7039a3fe3bd61d34
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251901
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36030}
2022-02-18 16:44:52 +00:00
d15f3e1220 Prepare the code to run ios tests with the standalone recipe.
The flags isolated-script-test-output and isolated-script-test-perf-output need to be consumed by the tests.

The generated .app folder in added in the data list of the gni file.
This will make it available in the runtime_deps file and thus will be populated to the swarming tasks.

Bug: webrtc:13556
Change-Id: I2c75774b847d9f686c3abc00ba0400bbc3fcefae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/240520
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Christoffer Jansson <jansson@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#36029}
2022-02-18 12:57:42 +00:00
18454b7720 Replace printf with RTC_LOG in YUV readers/writers
Bug: none
Change-Id: I70027a850b750067e0f7622fccfa724406974a1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251866
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36028}
2022-02-18 11:37:41 +00:00
29ff3efebf Reland: Make dcSCTP the default SCTP implementation
To disable dcSCTP and fallback to usrsctp, you can use the field trial
WebRTC-DataChannel-Dcsctp/Disabled/

Also remove a hidden no-break space in dcSCTP logging causing issues in
some log parsing.

Bug: chromium:1243702
Change-Id: I46136a8913a6d803a3c63c710f3ed29523e4d773
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251867
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Victor Boivie <boivie@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36027}
2022-02-17 23:57:01 +00:00
962bf18961 Use internal() in VideoTrack when invoking the source.
This skips going through the proxy and potentially hide a thread hop
should a regression occur.

This CL contains a part of a previously reviewed, landed, reverted,
relanded and re-reverted CL:
https://webrtc-review.googlesource.com/c/src/+/250180

Bug: webrtc:13540
Change-Id: If098f5c04a263547fb53f44e9f9738b8e941a294
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251861
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36026}
2022-02-17 14:25:50 +00:00
4bc7223cf0 Move VideoTrackSourceProxy creation into VideoTrack.
This CL contains a part of a previously reviewed, landed, reverted,
relanded and re-reverted CL:
https://webrtc-review.googlesource.com/c/src/+/250180

Bug: webrtc:13540
Change-Id: Id6df8da5ff61e3ec90d0b3f3d828e8f670d4931b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251860
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36025}
2022-02-17 13:44:51 +00:00
a6bab608df Report encode/decode latency
Bug: none
Change-Id: If36ee02ee762718b1c1b6f84cd22cb866ba0d51b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251863
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36024}
2022-02-17 12:17:42 +00:00
380d60c89a Delete INFO/WARNING/LERROR log levels
These have been depreated since November 2021.

Bug: webrtc:13362
Change-Id: Ifc1b984ab54faefc974006f37f909e6927aed056
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251760
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36023}
2022-02-17 10:51:12 +00:00
88f3c830f6 Revert "Make dcSCTP the default SCTP implementation"
This reverts commit 035e97a447455bdd434b6d775b824d3ee2bb2c8c.

Reason for revert: Breaks downstream projects.

Original change's description:
> Make dcSCTP the default SCTP implementation
>
> To disable dcSCTP and fallback to usrsctp, you can use the field trial
> WebRTC-DataChannel-Dcsctp/Disabled/
>
> Bug: chromium:1243702
> Change-Id: Ia90b796562245558a61481317bcded437400b045
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251800
> Auto-Submit: Florent Castelli <orphis@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36018}

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

Change-Id: Id4767920abc9a4d934b5d9bb49ea0b2178df950b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1243702
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251862
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#36022}
2022-02-17 09:01:56 +00:00
fe328ca88a Add several thread checks to RtpSender classes.
Minor related updates to AudioTrack and VideoTrack's sequence checkers.

There's more that can be done (or arguably needs to), but this is
a start.

Bug: none
Change-Id: I3ccf8eb9bbb6bef62b83248a23a68871b9fcd9e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251843
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36021}
2022-02-17 09:00:54 +00:00
27d5f14cf2 in RTPSender disallow enabling misconfigured rtx
Bug: None
Change-Id: Id94771626ef723212e4d92d9093af3ec9e647990
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251780
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36020}
2022-02-16 16:08:40 +00:00
419344264c Disable periodic keyframes
Bug: none
Change-Id: I8bd049cb8e8c958e59bf90a61198b7933eb5d40f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251692
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36019}
2022-02-16 15:35:01 +00:00
035e97a447 Make dcSCTP the default SCTP implementation
To disable dcSCTP and fallback to usrsctp, you can use the field trial
WebRTC-DataChannel-Dcsctp/Disabled/

Bug: chromium:1243702
Change-Id: Ia90b796562245558a61481317bcded437400b045
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251800
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36018}
2022-02-16 14:30:31 +00:00
df2b264ac0 [PCLF] Remove deprecated APIs
Bug: b/213863770
Change-Id: I69c0a9983831b7d59e24dc800a5f0d198cb40747
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251820
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/main@{#36017}
2022-02-16 14:23:59 +00:00
e6106102f8 Fix fd leak in ifaddrs_android.cc
allow absl::Cleanup for such purpose

Bug: webrtc:13674
Change-Id: I7434c7a48f1135bf4bf14b66996fbff1a7016c74
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251781
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36016}
2022-02-16 12:37:35 +00:00
44fd6e35d3 Return name of underlaying HW codec
Bug: none
Change-Id: I2c6943b91a6d58b884270a029bcf210e4d5c7a91
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251782
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36015}
2022-02-16 12:21:53 +00:00
b903e3b185 Use isHardwareAccelerated on Q+
Bug: none
Change-Id: Ic3f478c5cde12e8a4d1d121749c0414b254a3ea3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251695
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36014}
2022-02-16 10:43:42 +00:00
454d2309de Add bitrate adaptation tests
Bug: none
Change-Id: I3e2c503efc7a85a3daaa40cd8118c1b02d3b81cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251680
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36013}
2022-02-16 10:25:13 +00:00
1db0a261ca Reland "Reland "Remove unused APM voice activity detection sub-module""
This reverts commit 09aaf6f7bcfb4da644bd86c76896a04a41f776e1.

Reason for revert: downstream fixed (see https://chromium-review.googlesource.com/c/chromium/src/+/3461371)

Original change's description:
> Revert "Reland "Remove unused APM voice activity detection sub-module""
>
> This reverts commit 54d1344d985b00d4d1580dd18057d4618c11ad1f.
>
> Reason for revert: Breaks chromium roll, see 
> https://ci.chromium.org/ui/p/chromium/builders/try/linux_chromium_tsan_rel_ng/1080583/overview
>
> https://chromium-review.googlesource.com/c/chromium/src/+/3461512
>
> Original change's description:
> > Reland "Remove unused APM voice activity detection sub-module"
> >
> > This reverts commit a751f167c68343f76528436defdbc61600a8d7b3.
> >
> > Reason for revert: dependency in a downstream project removed
> >
> > Original change's description:
> > > Revert "Remove unused APM voice activity detection sub-module"
> > >
> > > This reverts commit b4e06d032e6f82a65c52ed0c5364ae9e7c0a0215.
> > >
> > > Reason for revert: breaking downstream projects
> > >
> > > Original change's description:
> > > > Remove unused APM voice activity detection sub-module
> > > >
> > > > API changes:
> > > > - webrtc::AudioProcessing::Config::VoiceDetection removed
> > > > - webrtc::AudioProcessingStats::voice_detected deprecated
> > > > - cricket::AudioOptions::typing_detection deprecated
> > > > - webrtc::StatsReport::StatsValueName::
> > > >   kStatsValueNameTypingNoiseState deprecated
> > > >
> > > > PSA: https://groups.google.com/g/discuss-webrtc/c/7X6uwmJarE0
> > > >
> > > > Bug: webrtc:11226,webrtc:11292
> > > > Change-Id: I8d008b56708cf62961b9857ec052b59fda3b41bf
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250666
> > > > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > > > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> > > > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > > > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/main@{#35975}
> > >
> > > TBR=gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
> > >
> > > Change-Id: Iee01fdb874b4e0331277f3ffe60dacaabc3859a2
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Bug: webrtc:11226,webrtc:11292
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251600
> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > > Cr-Commit-Position: refs/heads/main@{#35977}
> >
> > # Not skipping CQ checks because this is a reland.
> >
> > Bug: webrtc:11226,webrtc:11292
> > Change-Id: I2fcbc5fdade16bfe6a0f0a02841a33a598d4f2ad
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251660
> > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#35984}
>
> TBR=mbonadei@webrtc.org,gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: Ib308a3af2dcce85a0074ef5a4680ccec3f82712f
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:11226,webrtc:11292
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251688
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Auto-Submit: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35990}

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

Bug: webrtc:11226,webrtc:11292
Change-Id: Idfda6a517027ad323caf44c526a88468e5b52b65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251762
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36012}
2022-02-16 08:41:30 +00:00
5ae9b260ff Implement MouseCursorMonitorPipeWire to track cursor changes separately
Current implementation has mouse cursor as part of the screen itself
which means that everytime a cursor changes location, we have to update
whole screen content, which brings unnecessary load overhead. Using our
own mouse cursor monitor implementation allows us to track only mouse
cursor changes and update them separately for much better performance.

Bug: webrtc:13429
Change-Id: I224e9145f0bc7e45eafe4490de160f2ad4c8b545
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244507
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36011}
2022-02-15 23:03:41 +00:00
dfd69c2210 Move VideoTrack's content_hint property to the signaling thread.
Bug: webrtc:13673, webrtc:13681
Change-Id: I06810338bf5e44665e4d005d35636e9a98b1bd0b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251684
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36010}
2022-02-15 21:55:59 +00:00
250c31df00 Move enabled() methods for VideoTrack over to signaling
Bug: webrtc:13680
Change-Id: I0faf0d03541fce2b93c03857e01e85588389ccd1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251720
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36009}
2022-02-15 21:51:15 +00:00
0c6e34ce5c Ensure PipeWire doesn't use a Null SourceId
This has mostly seemed to work fine until now; but there's a collision
happening in chromium where if the source is being shown in the Window
Picker it collides with the (also null) Dialog ID and is ignored. While
we could patch that code to not count Null as a collision, there's the
potential for other (future) code to simply ignore a capture source
that it thinks is Null.

Fixed: chromium:1295375
Change-Id: I4356084f0af97f4d56632938b0d9a24d327f7107
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251500
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Auto-Submit: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36008}
2022-02-15 20:03:33 +00:00
0b02d637c0 Calculate max/avg encode/decode latency in codec tests
Bug: none
Change-Id: Ie42461dd06b1764c99308393477921ea25319ab4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251687
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36007}
2022-02-15 18:14:41 +00:00
b92d3e6ef9 [PCLF] Move FEC and bitrate mulitplier into per peer configs
Bug: b/213863770
Change-Id: Idcf37150e769db18d4a12baa1057840d521b8e1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251761
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36006}
2022-02-15 18:10:31 +00:00
ac341df436 Adding fuzzer for PCM16b decoder and fixing a fuzzer problem
Bug: chromium:1280852
Change-Id: I7f6c5de86ceee01156743c0389c59f875e53bb5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251580
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36005}
2022-02-15 15:59:01 +00:00
e47493b3c0 Add restrictive visibility to all targets in //pc
This CL sets all visibility to ":*" (this buildfile) where no users
outside this directory are known, and marks up publicly exported
targets and Chrome dependencies explicitly.

Bug: webrtc:13661
Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 14:40:52 +00:00
1b083a998b Encode data for compression + add initial tests
Bug: webrtc:13607
Change-Id: I3bbec5558e676ca45125fad3fdbd10cc47c84601
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251390
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@{#36003}
2022-02-15 14:24:11 +00:00
23bb9d75fc Allow designated initializers in WebRTC
to align with chromium and google style guides

Bug: None
Change-Id: I92e1bb6d187eac6b531d495aedb8176f66186a5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251689
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36002}
2022-02-15 13:23:12 +00:00
599002c905 Restrict frame id range in frame buffer 3 fuzzer
Bug: chromium:1293129
Change-Id: Icc9152447363e69b2be561bc90a23f411d64b11a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251385
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36001}
2022-02-15 09:18:51 +00:00
987b671017 Add ability to add peer to the stats poller during the test
Bug: b/213863770
Change-Id: I65e0338806f808329725fce50d778738724cf13d
Pair: mbonadei@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251693
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36000}
2022-02-15 09:09:02 +00:00
f7a1937e70 Add FrameBufferProxy test for low-latency renderer
Ensures that frames are decoded instantly when in low-latency render
mode. This also tests the max queue size behaviour. Adds a new test
suite for FrameBufferProxy that sets the appropriate field trials.

* Fixes FrameDecodeTiming to never use negative wait times for decode
timestamps.

R=kron@webrtc.org

Change-Id: I06cbec52e1e866e21aa964b24c4fd0163c26961b
Bug: webrtc:13658
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251601
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35999}
2022-02-15 08:30:51 +00:00
405ac4e840 Add objc_class_prefix to the Audio Network Adaptor proto.
WANA: WebRTC Audio Network Adaptor

No-Try: True
Bug: None
Change-Id: I291e02ab70323ecc45d87cea0ea8d7e8cb62db9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249784
Reviewed-by: Minyue Li <minyue@google.com>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35998}
2022-02-14 21:04:20 +00:00