The const-ref result of .str() must be copied into the returned
value, whereas the result of .Release() can be moved.
Bug: webrtc:374845009
Change-Id: I3abc98be30ce9947127c7664f5ffa6846b772ea2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366480
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43288}
For now, old file forward include api/transport/ecn_marking.h
Done in preparation for more usage of this enum when handling received
RFC8888 feedback.
Bug: webrtc:42225697
Change-Id: I022c2b7f1e7f986b24aa32b8911ad67c6640a5c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366440
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43282}
This would simplify migrating from PeerConnectionFactoryDependencies::audio_processing
for users who use own implementation of the AudioProcessing
Bug: webrtc:369904700
Change-Id: Id05f7280fd01a3e8fd4953f1b24b2467335ab065
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366120
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43273}
This would allow users of the voip engine to migrate away from the AudioProcessingBuilder
Bug: webrtc:369904700
Change-Id: Ie4f6f4579e185ff6366333a3f37e6aaa23b892b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365920
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43255}
It would allow to use EncodedImageBufferInterface with gtest container matchers.
Bug: None
Change-Id: Iae37d1a019e044a4ec583c32e8141fe0758e60ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365501
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43253}
Compiling webrtc with `-Werror=unused-parameters` is failling duo to
those parameters.
Also, it shouldn't harm us to put those in comment for code readability as
well.
NOTE: This time I made sure to iterate over the C files in the
audio_processing folder and compile them using gcc.
On the original CL that was reverted - that failed with the same error
Danil mentioned. This time it seems fine.
I'll make sure to run the same script on the rest of my CLs for sanity
Bug: webrtc:370878648
Change-Id: I83cea3a08777e21d26a95bcad503a2d1b74566eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364537
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
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@{#43249}
Its implementation is a copy of the AudioProcessingBuilder with intention to replace all usage of AudioProcessingBuilder with the BuiltingAudioProcessingFactory and thus get Environment with propagated field trials available for AudioProcessingImpl at construction.
Bug: webrtc:369904700
Change-Id: Iee0eb112dd579402fcd5be56bf1054946179d1fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365582
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43242}
This interface allows to delegate construction of AudioProcessing to
the PeerConnectionFactory where it can provide propagated field trials
Bug: webrtc:369904700
Change-Id: Ie05cd771e4a869fa5f43173e127256800ae0727f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365320
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43233}
The goal is to be able to write the rtc::Buffer by another utility
(like rtc::ByteBufferWriter) and pass it into EncodedImageBuffer
without memcpy.
Bug: webrtc:42223344
Change-Id: Ieda55e77a36636e8cdff6ad6b7d078de0aeafec0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364243
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43179}
To support libraries and dependencies compatible with absl way of debug printing custom types.
In particular gtest can use AbslStringify to produce nice output when unit types are compared with EXPECT macros.
Bug: None
Change-Id: Ie78293a225f61977f256f0234e07d166b1977e2d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364162
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43164}
Compiling webrtc with `-Werror=unused-parameters` is failling duo to
those parameters.
Also, it shouldn't harm us to put those in comment for code readability as
well.
Bug: webrtc:370878648
Change-Id: I0ab2eafd26e46312e4595f302b92006c9e23d5d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364340
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43157}
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 f8b3dab7c6320a9890f0b003b43d7099e2e00a5b.
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}
