Add media config for using environment monotonic timestamps (i.e. not UTC) in RTCStats constructor, and implemented the usage of the flag.
Bug: chromium:369369568
Change-Id: Ia93d048742c28af201164fe7b2152b791bb6d0b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363946
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#43156}
These have been deprecated since 2022.
Bug: None
Change-Id: I8340750f67e57c37601754345c679062c3c23436
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364283
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43122}
This reverts commit f8b3dab7c6.
Reason for revert: The fix landed in libaom (https://aomedia-review.googlesource.com/c/aom/+/193761) and it is now available in WebRTC (import CL: https://webrtc-review.googlesource.com/c/src/+/364126).
Original change's description:
> Disable LibaomAv1Encoder tests to unblock Chromium roll
>
> The tests exercise the new encoder API that is not used in prod yet.
>
> Bug: webrtc:369633254
> Change-Id: Iee6bc16ebd471f4accdd9531cdb404f159557f51
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363820
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43083}
Bug: webrtc:369633254
Change-Id: Ia02db32f7f09e3abc3d0a46605feeabd82673f06
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364281
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#43120}
These functions have been deprecated since October 2022.
Bug: None
Change-Id: I74f51c9d0e8ee340a2043bf43f7a1b0d8b79726e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364280
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43118}
By this change we aim to remove the flag enable-webrtc-srtp-encrypted-headers.
Bug: chromium:40623740
Change-Id: I74692c90ff1caf2a11d7b73211c1ae4472edfb4d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362740
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43105}
The tests exercise the new encoder API that is not used in prod yet.
Bug: webrtc:369633254
Change-Id: Iee6bc16ebd471f4accdd9531cdb404f159557f51
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363820
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43083}
Level asymmetry is implicitly enabled for HEVC. When comparing two
codec params to see if they match, we only compare profile & tier,
similar as H.264.
Bug: chromium:41480904
Change-Id: I9e9debdf1b34f33986da9344b9fee14071b1ed60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363205
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#43069}
This is a follow up for https://webrtc-review.googlesource.com/c/src/+/360680.
* Adding some missing <optional> include.
* Adding a IWYU pragma to force keeping an include.
Note that I've added the CQ bot 'iwyu_verifier' to ensure the repo stays clean. It is still work in progress and it currently needs to be triggered manually.
FYI I used these command line to run iwyu:
> for i in api/*.cc; do ./tools_webrtc/iwyu/apply-include-cleaner $i; done
> for i in api/*.h; do ./tools_webrtc/iwyu/apply-include-cleaner $i; done
Change-Id: Ie7036d08edbb6884f2b35eb9d69646757d662390
Bug: webrtc:42226242
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362440
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#43054}
The `requested_resolution` API must not change aspect ratio, example:
- Frame is 60x30
- Requested is 30x30
- We expect 30x15 (not 30x30!) as to maintain aspect ratio.
This bug was previously fixed by making VideoAdapter unaware of the
requested resolution behind a flag: this seemed OK since the
VideoStreamEncoder ultimately decides the resolution, whether or not
the incoming frame is adapted.
But this is not desired for some non-Chrome use cases. This CL attempts
to make both Chrome and non-Chrome use cases happy by implementing the
aspect ratio preserving restriction inside VideoAdapter too.
This allows us to get rid of the "use_standard_requested_resolution"
flag and change the "VideoStreamEncoderResolutionTest" TEST_P to
TEST_F.
Bug: webrtc:366067962, webrtc:366284861
Change-Id: I1dfd10963274c5fdfd18d0f4443b2f209d2e9a4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362720
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43037}
This will help to reduce redundant ScalabilityMode to temporal layer
count mapping in blink.
Bug: chromium:40763991
Change-Id: Ida3e6abb91383e27465eb1b697ad9431935cf9ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362486
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#43031}
When requested_resolution uses a different aspect ratio than the source
the encoder will restrict the frame without changing its aspect ratio,
e.g. a 60x30 input frame that is restricted to 30x30 results in 30x15,
not 30x30.
While this logic works correctly in isolation, if the source also adapts
the frame size based on the sink_wants.requested_resolution that is
signaled back to the source, then the source will produce stretched
30x30 prior to the encoder which happily sends 30x30 not knowing any
wiser.
This is incompatible with the spec[1] and makes this WPT[2] fail. The
correct behavior is to NOT signal the requested_resolution back to the
source, the encoder already configures the correct resolution so this
isn't actually needed and the source shouldn't need to know this API.
In order not to break downstream projects, the new behavior is landed
behind a flag and both behaviors are tested with TEST_P.
This unblocks launching scaleResolutionDownTo API on Web. Migrating
from old to new code path and deleting the flag is a follow-up AI:
webrtc:366284861.
[1] https://w3c.github.io/webrtc-extensions/#dom-rtcrtpencodingparameters-scaleresolutiondownto
[2] https://chromium-review.googlesource.com/c/chromium/src/+/5853944
# Relying on previous green runs for confidence due to purple bots atm,
# see b/367211396
NOTRY=True
NOPRESUBMIT=True
Bug: webrtc:366067962, webrtc:366284861
Change-Id: I7fd1016e9cc6f0b0b9b8c23b0708e521f8e12642
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362541
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43024}
* IWYU export <sys/socket.h> from rtc_base/net_helpers.h.
* Add a presubmit check to ensures that <sys/socket.h> is included through net_helpers.h (expect if there is a IWYU pragma or a no-presubmit-check).
* Clean up existing includes of <sys/socket.h>
Change-Id: I4bc6cce045c046287f8f74f89edfc9321293b274
Bug: b/236227627
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362082
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42996}
Now some HW encoders support simulcast. If parameters are not suitable for
single encoder simulcast, the error code should be forwarded back to
SimulcastEncoderAdapter instead of trying software fallback.
Bug: webrtc:347737882
Change-Id: Id02ff1afc012cd46761d9530b1ce368d5dc480bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361744
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42972}
Remove mention of absl_deps - it is history already.
Rewrite motiviation of banning absl::Span to be up to date with c++20 state.
Remove motivation of banning absl::Mutex as it likely no longer accurate, and that ban might be re-evaluated.
Ensure allow list matches what is in root DEPS
No-Try: True
Bug: b/363943024, webrtc:342905193
Change-Id: I890a87511bafac7c51355d8f49e0237352eee7b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361302
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42914}
Moving defines before they are used with
unique_ptr allows to compile this file with
-std=c++2b flag.
Bug: webrtc:339074792
Change-Id: Ie7c37ab724800aea4545b72b4d2a779e12a2026a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360860
Auto-Submit: Ho Cheung <hocheung@chromium.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42879}
This is currenly tracked in both AcmReceiver and NetEq. Adding this API
enables us to have it in just one place.
Bug: None
Change-Id: Ia537f87f36b0aedf19c00a57bd6cec4425a49df1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360743
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Cr-Commit-Position: refs/heads/main@{#42872}
The "at target quality" attribute is no longer set to the encoded
image in VideoStreamEncoder, see
https://webrtc-review.googlesource.com/c/src/+/359640
Mark the attribute as deprecated to avoid new dependencies and prepare
for deletion.
Bug: chromium:359410061
Change-Id: Id5a98ec9d2068099cb75a70b849bbf1c77feabb3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359660
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42848}
This allows these types to meet the requirements of e.g.
std::ranges::range, which is necessary for them to work with the
std::span range constructor, or the "non-legacy" constructor for
Chromium's base::span.
Bug: none
Change-Id: Ia51c17690c785e0714c36d237094877129e0cbaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358844
Auto-Submit: Peter Kasting <pkasting@chromium.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42838}
and update some usage to use the "correct" stun attribute names
BUG=webrtc:42229250
Change-Id: If0c34d1d9b399766d7073661ea2a5515100256a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359440
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42810}
Needed to get rid of a form of CreateSessionDescription that is due
for deprecation.
Bug: None
Change-Id: I9717b7ded1e28cf803de4bebc852c2f255851918
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359941
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42808}
Mark old overload deprecated.
This allows to migrate both calls through AudioDecoderFactory and direct calls to AudioDecpderOpus trait.
Bug: webrtc:356878416
Change-Id: I1502aee5b18aac43a8258e77b770c8e73a056f92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359741
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42793}
Finalize change started in https://webrtc-review.googlesource.com/c/src/+/359243
Remove fallback to old interface and unneeded clock member in the config struct.
Bug: None
Change-Id: I4c2b65a09dd1c8a0d44ee76320b095516e2c61fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359561
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42782}
e.g all files in the api/test folder not including subdirectories
Bug: webrtc:42226242
Change-Id: I18d74a18f8feec41eb252faa9acfffd1d6f45ce4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Cr-Commit-Position: refs/heads/main@{#42773}
