(skip-generation)
Upstream commit: https://webrtc.googlesource.com/src/+/cd09858f4a3f45e2c60508ababac6031986cdf7d
Convert decoder TRACE_EVENT to flows
This is the first new style trace event so this CL adds and updates
WebRTCs Perfetto configuration.
* Changes all #includes to target "third_party/perfetto". Added this
to DEPS.
* Expose the Perfetto public config in the "tracing" group using
an all_dependent_configs statement. This means the config is included
in all parts that include the "//:tracing" group. However, direct
perfetto includes are banned per DEPS.
* In order to expose Perfetto types (ie Flow/TerminatingFlow) the
perfetto headers are a dependancy on all targets. This should not
affect binary size as these are not used when perfetto is not enabled
and will not be linked.
Bug: b/42226290
Change-Id: I5711d81dae95ee907cbcd41bf1ee9b55d2ec595c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349161
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#42197}
Upstream commit: https://webrtc.googlesource.com/src/+/db50b0355311703b4ba13a866883e7f0e5717ee5
Add perfetto build config
This adds Perfetto support to WebRTC with a GN flag rtc_use_perfetto.
The configuration of perfetto depends on whether or not webrtc is
build within Chrome or not. When in Chrome, WebRTC will depend on
//third_party/perfetto:libperfetto. When building standalone, specific includes required for Perfetto are exposed with the library webrtc_libperfetto.
The perfetto trace API is exposed with a header export in
trace_event.h which is used instead of the legacy API.
The addition of Perfetto means there are 4 compilation modes for
tracing in WebRTC,
1. No tracing implementation.
2. Legacy tracing (AddTraceEvent/GetCategoryEnabled).
3.a. Perfetto statically linked (webrtc_libperfetto).
3.b. Perfetto in Chrome (Chrome's libperfetto).
This CL removes the tracing expectations from
rtc_stats_integrationtest.cc because those directly used the old API.
Integration into Chrome is a follow up CL which depends on
https://chromium-review.googlesource.com/c/chromium/src/+/5471691.
Tested: Ran Chrome with Perfetto and traces appear. WebRTC Unit test tracing working: https://ui.perfetto.dev/#!?s=04ea2613ea36b814394639a1ec4b60be5b5097527f1a485995ecc13469885468
Bug: webrtc:15917
Change-Id: I537d79dc247c2b759689910c621087286a4d8fdc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347880
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@google.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mikhail Khokhlov <khokhlov@google.com>
Cr-Commit-Position: refs/heads/main@{#42166}
Upstream commit: https://webrtc.googlesource.com/src/+/9fdceb80b5f7c36cdbcd5861c737b40a73b430f6
Add environment_construction poison
This poison guards against accidental use of EnvironmentFactory and thus ensures low level WebRTC class would use utilities from propagated environment instead of accidentally using a default implementation.
This poison extends and thus replaces default task queue poison.
Bug: webrtc:15656
Change-Id: I577bef8af08b9c7dd649ad5a2284eb236e6f4a8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328380
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41247}
Upstream commit: https://webrtc.googlesource.com/src/+/2e1f16d55c66ded230db084cdd517a9b9dfb21bc
Make AEC3 json parsing code testonly
Reasons:
- the code is no longer used in Chrome
- it is conceptually weird for WebRTC to have JSON parsing in its API
- there are concerns around the reliability of the underlying JSON library
Additionally, this CL removes the rtc_json "poisonous" attribute: the scheme is incompatible and redundant with testonly.
Bug: webrtc:1493351
Change-Id: I0b621b0e3f183df7315919d9c89242fbe387928f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325062
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41014}
Upstream commit: https://webrtc.googlesource.com/src/+/f2827c4b1a7eb574fd83f208001a75b880205c01
FrameCadenceAdapter: schedule repeats before issuing decodes.
The code currently issues frames for encode before scheduling
a new repeat. Swap this order to account for time taken by for
slow encodes.
Bug: webrtc:15456
Change-Id: I74177069e30c1bf65268231ffba033411a0f7b9a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318580
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40690}
Upstream commit: https://webrtc.googlesource.com/src/+/9130431b54fd18aa4857614c886aeeed1731fcf7
Add possibility to set RTC_OBJC_TYPE_PREFIX from GN.
This CL also adds the prefix RTC_TESTING to `ios_internal_pure_release_bot_arm64` in order to avoid ODR
violations.
Bug: b/292472934
Change-Id: If63020e679c8670b4c797217eb38fc8c2954d422
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313240
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40476}
Upstream commit: https://webrtc.googlesource.com/src/+/1e6d77c29afb562ac8b839b9f19b9b576419bb32
Always check out google_benchmark, part 3.
Add a WebRTC-specific arg that can be used to control use of targets
that rely on //third_party/google_benchmarks, so the .gni in that
directory can eventually be removed.
Bug: chromium:1404759
Change-Id: I2a9422fae119ca13eb50028d962fc0a671b5fb33
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297460
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Peter Kasting <pkasting@chromium.org>
Cr-Commit-Position: refs/heads/main@{#39553}
Upstream commit: https://webrtc.googlesource.com/src/+/ff16f5f88e1af6b553abe2a847436ef0adce0900
Add a new test binary to check if split-channel works
This test (and binary) will be used to verify that the field trial
for enabling split-MediaChannel works in both "off" and "on" modes,
so that it can be run as a field trial. It is intended to be deleted
once the conversion to split-MediaChannel is complete.
Bug: webrtc:13931
Change-Id: If62d19be9b2f205067b86dc859946279442fdd58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296322
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39493}
Upstream commit: https://webrtc.googlesource.com/src/+/9109e856d53c4e18961e20b72cea1d23dc70408f
Add option to log a warning for unregistered field trials
Until now you only had the option to RTC_DCHECK for unregistered field
trials. This makes it possible to log a warning instead.
Bug: webrtc:14154
Change-Id: I8628054e3c9b5d690f241a93e61299126b732ed0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295300
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39417}
Upstream commit: https://webrtc.googlesource.com/src/+/6bf20cc76a59e1f548ca53c6ef21f4143e3fe974
Verify field trials looked up through field_trial::FindFullName
For now, the run-time check will only be enabled if the
rtc_strict_field_trials GN arg is set.
In order to allow testing with imaginary field trial keys, two test
helpers have been added. It's a bit awkward to test these since the
field trial string is already global, hence the helpers are also
modifying global state. Tests must make sure this global state is reset
between runs. Things won't be an issue anymore when [1] has removed the
global string.
[1] https://crbug.com/webrtc/10335
Bug: webrtc:14154
Change-Id: Ida44cc817079d7177325e2228cf1f1d242b799e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276269
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38447}
Upstream commit: https://webrtc.googlesource.com/src/+/1264dc165b9989e3e53eddbefe054c6e86bbf1f7
PipeWire capturer: add initial test for SharedScreenCastStream
This test created another PipeWire stream we can connect to with
SharedScreenCastStream and recieve frames from there. This is an
initial version, where I test whether we can successfuly connect
and disconnect, receive frames and it also tests DesktopFrameQueue.
In the future I will add tests to test mouse cursor and try to
come up with some corner cases and possible scenarios.
Bug: webrtc:13429
Change-Id: Ib2a749207085c6324ffe3d5cc8f2f9c631fa6459
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256267
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38431}
Upstream commit: https://webrtc.googlesource.com/src/+/bcf24f5bcdc02e984859d15fd1a0d3106be08c41
Move TimestampExtrapolator closer to its single user
The `TimestampExtrapolator` is only used by the `VCMTiming`
class, despite there being references to it from both
`modules/rtp_rtcp/BUILD.gn` and `modules/video_coding/BUILD.gn`.
Bug: webrtc:14111
Change-Id: If1a02a56a0c83b13d619ca08dc76c884fa829369
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275482
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38093}
The following git commits have had history rewritten to correct issues we created as
we moved quickly through getting libwebrtc builds working. In a few cases, the
commit disappears in our github stack because a change could be reverted completely.
Bug 1654112 - Tweak upstream gn files for Firefox build. r=ng
Bug 1654112 - Changes to support building with external libyuv. r=ng
Bug 1766646 - (fix-f9ee0e08ec) limit create_cross_traffic deps
Bug 1766646 - (fix-15a3c3fdca) restrict deps in windows_version_unittest
Bug 1766646 - (fix-8615bf0582) restrict deps in frame_buffer_unittest
Bug 1766646 - (fix-b6f002b55f) build fixes for new usage of libyuv
Bug 1766646 - (fix-5cfcf2282a) build fixes for new usage of libyuv
Bug 1766646 - (fix-b09d87232b) fixes for new usage of libyuv and libdav1d
Bug 1766646 - (fix) breakout Call::Stats and SharedModuleThread into seperate files
Bug 1766646 - forking base_capturer_pipewire.cc before e1e05afec7d7d10bb77fd00c5bd72cf17d77120d
Bug 1790097 - (fix-2a2f3ece15) remove build DEP for upstream's third_party/libaom
Bug 1790097 - (fix-4eab9f47a5) remove build DEP for upstream's third_party/libaom
Differential Revision: https://phabricator.services.mozilla.com/D163992
Upstream commit: https://webrtc.googlesource.com/src/+/90b74389a2b4210036960f2e6af75c86518ac198
SVC: Add end to end tests for VP8 and VP9
The tests check that the various scalability mode are supported
and the frames are marked properly by the encoder with their
spatial and temporal index.
The same information is then checked on the receiving side.
A new member is added on EncodedImage to store the temporal index,
and is filled by the encoders and retreived by the ref finder
objects on the decoding side.
Bug: webrtc:11607
Change-Id: I7522f6a6fc5402244cab0c4c64b544ce09bc5204
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260189
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37303}
Michael Froman
Dan Baker
Jan-Ivar Bruaroey
Byron Campen
Andreas Pehrson