To stress there is no intention to use each instance more than once.
Bug: webrtc:369904700
Change-Id: Id53ad804f39f8ee596ec0b45ff15393009fdfab0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366640
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43324}
It is used to distinguish between audio/video packets and everything else (retransmit/padding/fec), so naming it is_media makes more sense.
This is a follow up to https://webrtc-review.googlesource.com/366644
Bug: b/375148360
Change-Id: Ia53f4d707ceb85f059688d86bc5dcc2d57908d88
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366424
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43319}
After landing this change, we can change the corresponding usage in
blink to start using presentation_timestamp as well and then delete
the remaining usage of capture_time_identifier.
Bug: webrtc:373365537
Change-Id: I0c4f2b6b3822df42d6e3387df2c243c3684d8a41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365640
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#43317}
Now that Chromium and internal_compile_lite have migrated we can delete
the old name.
Bug: webrtc:375048799
Change-Id: I11d79f1d4ef1c0aa132cb50856faf83250e07caf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366600
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43312}
It works in the same way as the first packet received callback and can be used for latency measurements.
One important detail is that RTCP and probe packets are excluded from triggering the callback.
Bug: b/375148360
Change-Id: I5f99b565f96b622e864669cf227be5534aab0fc7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366644
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43309}
Removing AudioProcessingBuilder from few layers would simplify replacing with BuiltinAudioProcessingFactory in the upcoming patches.
While doing cleanup also removed extra always empty parameters and run iwyu.
Bug: webrtc:369904700
Change-Id: I54d44993701c30ca8f4cf38e822af08531fba310
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366260
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43306}
This is a reland of commit 82617ac51e7825db53451818f4d1ad52b69761fd
The reason for the revert was a downstream use of
`rtc::VideoSinkWants::requested_resolution`, so in this reland we don't
rename this field, it's fine just to rename the one in
RtpEncodingParameters for now.
Original change's description:
> Rename `requested_resolution` to `scale_resolution_down_to`.
>
> This is a pure refactor/rename CL without any changes in behavior.
>
> This field is called scaleResolutionDownTo in the spec and JavaScript.
> Let's make C++ match to avoid confusion.
>
> In order not to break downstream during the transition a variable with
> the old name being a pure reference to the renamed attribute is added.
> This means we have to add custom constructors, but we can change this
> back to "= default" when the transition is completed, which should only
> be a couple of CLs away.
>
> Bug: webrtc:375048799
> Change-Id: If755102ccd79d46020ce5b33acd3dfcc29799e47
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366560
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43300}
NOTRY=True
Bug: webrtc:375048799
Change-Id: Ic4ee156c1d50aa36070a8d84059870791dcbbe5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366660
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43304}
This reverts commit 82617ac51e7825db53451818f4d1ad52b69761fd.
Reason for revert: Break downstream projects
Original change's description:
> Rename `requested_resolution` to `scale_resolution_down_to`.
>
> This is a pure refactor/rename CL without any changes in behavior.
>
> This field is called scaleResolutionDownTo in the spec and JavaScript.
> Let's make C++ match to avoid confusion.
>
> In order not to break downstream during the transition a variable with
> the old name being a pure reference to the renamed attribute is added.
> This means we have to add custom constructors, but we can change this
> back to "= default" when the transition is completed, which should only
> be a couple of CLs away.
>
> Bug: webrtc:375048799
> Change-Id: If755102ccd79d46020ce5b33acd3dfcc29799e47
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366560
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43300}
Bug: webrtc:375048799
Change-Id: Ie41723a39420e12e7b5b681d3d00ccd14f66b4b1
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366642
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#43301}
This is a pure refactor/rename CL without any changes in behavior.
This field is called scaleResolutionDownTo in the spec and JavaScript.
Let's make C++ match to avoid confusion.
In order not to break downstream during the transition a variable with
the old name being a pure reference to the renamed attribute is added.
This means we have to add custom constructors, but we can change this
back to "= default" when the transition is completed, which should only
be a couple of CLs away.
Bug: webrtc:375048799
Change-Id: If755102ccd79d46020ce5b33acd3dfcc29799e47
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366560
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43300}
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}
