Update to use similar bitrate limits as VP9, depending on whether QP
from encoder is trusted or not.
Bug: chromium:392060821
Change-Id: I305182fecf7acfe84dbd2e049f9ce712a7a30ede
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/376762
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43898}
std::aligned_storage is deprecated in C++23, see linked bug. The only
webrtc usage (VoidUnion::inline_storage) employed the default Align
parameter (i.e. std::aligned_storage<Len>, not std::aligned_storage<Len,
Alignment>), which is defined as the largest alignment <= Len.
This patch replaces the usage with a char buffer of same size and
alignment alignof(std::max_align_t). In theory, this might increase
alignment and use more memory. In practice, local testing in my
machines showed no behavior change, since Len =
kInlineStorageWords * sizeof(uintptr_t) = 32 > 16 (maximum alignment
on linux x86_64) > 8 (maximum alignment on mac arm64).
Bug: chromium:388068052
Change-Id: If08b69f7a5ff7b716a66c8703e31eb5d4de14431
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/373800
Commit-Queue: Victor Vianna <victorvianna@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Victor Vianna <victorvianna@google.com>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43842}
This CL focuses on addressing the remaining TODO items in the
`//rtc_base/bounded_inline_vector_impl.h` file, mainly involving the two
functions `void DefaultInitializeElements` and `void MoveElements`.
In the `void DefaultInitializeElements` function, the placement new usage is
adopted. When using placement new, manual construction operations on objects are
usually required, and a fair amount of logic needs to be written to ensure
correct object construction. In contrast, the
`std::uninitialized_default_construct_n()` function only requires passing in the
starting address of the memory and the number of objects to be constructed, and
it can automatically construct objects in batches.
In the `void MoveElements` function, the original implementation also uses the
placement new usage. When using placement new to move objects, manual handling
of the moving operation for each object is required, and usually, `std::move`
needs to be combined to achieve the object's move semantics. However, the
`std::uninitialized_move_n()` function can automatically complete the object
moving process just by passing in specific parameters.
To improve the code's simplicity, it is considered to use
`std::uninitialized_default_construct_n()` and `std::uninitialized_move_n()` to
replace the original two placement new usages.
Fixed: webrtc:392037564
Change-Id: If8edcd9ac8a4d85be0ce0a23f6433d7f91b39ad3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375182
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Ho Cheung <hocheung@chromium.org>
Cr-Commit-Position: refs/heads/main@{#43831}
This was already done in one place but got caught by our linter
nonetheless. For better obfuscation split "PRIVATE" into two pieces.
BUG=None
No-Iwyu: mostly unrelated changes and some require special attention
Change-Id: Iba82b603fd5c5a50c75fc7e27cafbc7237e956f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375063
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43798}
Drive-by: fix MakeVal in rtc_base/checks.h to ensure that StrCat is
used for types that have AbslStringify.
Bug: None
Change-Id: Ia78c65da18b4a826365a6a2c741f11809640197f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374345
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43779}
which shipped in M131.
BUG=webrtc:367181089
Change-Id: I967bafbab99bf211b0e6a4454f6598fd52493b26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370080
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43674}
This patch fixes a scenario in which the ssl_ object was freed.
BUG=webrtc:383141571
Change-Id: I413b028473aff9c3078f9bbab8e1fee718623417
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/372340
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43621}
This patchs adds a field trial for enabling DTLS1.3, WebRTC-ForceDtls13
- "Enabled" set max version to DTLS1.3
- "Only" set min & max version to DTLS1.3
Wireup a FieldTrialsView so that this does not
use the global string.
Also convert the WebRTC-DisableTlsSessionTicketKillswitch
from global string to FieldTrialsView.
BUG=webrtc:383141571
Change-Id: Ia775efc1dcbffd01bfddb6030490438cb8de89d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/372261
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43616}
Modify openssl_stream_adapter to check/set
timer regardless of dtls state. This is needed for DTLS1.3
orelse handshake will never complete if last client
packet is lost (e.g if retransmit is not triggered after
writable) as show by TestHandshakeLoseSecondClientPacket.
TestHandshakeLoseSecondClientPacket works with/without this
patch if using DTLS1.2.
BUG=webrtc:383141571
Change-Id: I2757783c9e79686d1fbe0eff12341ab9e3863fdd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/372201
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43610}
- add DTLS1.3 ciphers (without KeyType)
- remove code in dtls_transport.cc that tries to parse DTLS packet
- cleanup some test
- start on test for packet loss during dtls handshake (more to come!)
After this patch is submitted, it is possible
to set max version = dtls1.3 and it will active
but DON'T do it yet.
BUG=webrtc:383141571
Change-Id: I6f9a120c53415ccee7a560ea83bd0c2636702997
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371300
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43595}
This allows other tests using RTC stats to get pretty printing as well.
Bug: webrtc:381524905
Change-Id: Ib1eb9e1dad36b89e5b1c2ec687fcfeb308f82939
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370761
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43575}
Previous CLs that disabled the rtc_enable_libevent build flag
did not reveal issues. Now continue to remove the source code for
the task queue.
Bug: webrtc:42224654
Change-Id: I0866b4b56f0a8d8b56a5b604c31a426d77ab8d04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370801
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43550}
Add SSLProtocolVersion for TLS13 and DTLS13
Allow setting max version to 13 (for BoringSSL)
Don't change any defaults.
This is a NOP.
BUG=webrtc:383141571
Change-Id: I11303c14e8d79c09d9437d44e44003c67d2fc31b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370900
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43530}
and include explanation of source access.
Bug: chromium:362397798
Change-Id: I7af673ffe060507b0e9dea95d650ffb0a681727c
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370120
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43484}
When an enum has AbslStringify, we log the text coming from stringifying it, not the numeric value.
Drive by changes,
1. Changed the tests to use string matchers rather than
std::string::find.
2. Fixed test includes.
3. Fix spelling.
Bug: webrtc:381502973
Change-Id: I6bab0afda1b20d72c02629e80ff2ac567650183a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369861
Auto-Submit: Evan Shrubsole <eshr@google.com>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43471}
Return default value T() if no values have been added to the filter.
Together with
https://webrtc-review.googlesource.com/c/src/+/369440, DataRate etc can be used by the filter.
Bug: None
Change-Id: I3d0e1a3e698a91a6197bf434ace2ff8246dc393e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369420
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43461}
0 is natural default value for types that can be accumulated
Having default constructor simplify usage of these types in templated code.
Bug: None
Change-Id: If005c69018a2a11011bc789502fdbc600cad3278
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369440
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43460}
When receiving streams with frame rates around 1 fps, the decode and
render fps were incorrectly reported as 0, even though frames were being
decoded successfully.
This commit addresses the issue by adjusting the calculation in
RateStatistics to better handle streams with frame intervals that are
close to the window size.
1 fps streams are an important special case that occur frequently in
in screen share scenarios.
Fixed: webrtc:354625675
Change-Id: I1362768229a3abab5929220ba4bbd5ccb06a33d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368080
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43417}
Instead use the preprocessor to avoid compiling Perfetto related code
when RTC_USE_PERFETTO is not defined.
Bug: None
Change-Id: I85b37cb0287327035ac2e8feb3caf9505486a1e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368343
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43410}
by going from cartesian product of combinations to an explicit
minimal list.
BUG=webrtc:375552698
Change-Id: I99b9afd7376f19abde54dafd8917954617d8c255
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367504
Reviewed-by: Christoffer Dewerin <jansson@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43373}
With L4S in WebRTC, only RTP packets are supposed to be send with ECT(1)
Bug: webrtc:42225697
Change-Id: If10bf74a867d3ea04fd1fb931cdc2a6380176270
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367220
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43343}
and modify DEPS files accordingly.
This is done in support of the decision to encourage AbslStringify.
Bug: None
Change-Id: I26fee77978d1dd21be6d2ef011c4dfd78a7b43e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367204
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43338}
and remove the testing of the nondefault key size from the "server" parameters to speed up tests
BUG=webrtc:375552698
Change-Id: Ibc1bd491300964aa45826b98962ed3e56c6d4974
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366941
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Christoffer Dewerin <jansson@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43321}
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}
Otherwise, the captured variables are immutable, so a subsequent
std::move() silently degrades to a copy.
-- Add missing () for consistency with other no-arg lambda
captures.
Bug: webrtc:374845009
Change-Id: I205589ff8047446918a45203a22620846b271187
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366280
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@{#43289}
Mentioned reasons for re-implementing it do not apply when simplest form of aliasing is used.
Bug: webrtc:42225969
Change-Id: If8b51f173faf4c66cde74413f4fbc7c72ad87323
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366460
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43285}
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}
and use uint8_t instead of unsigned char. Follow-up from
https://webrtc-review.googlesource.com/c/src/+/365274
BUG=webrtc:357776213
Change-Id: Ibc97e5cc85316ba69b4133b7f3c42e3afbdd7abd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365540
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43263}
This allows this type to meet the requirements of e.g.
std::ranges::range, which is necessary for it to work with the std::span
range constructor, or the "non-legacy" constructor for Chromium's
base::span.
Bug: chromium:364987728
Change-Id: I6cb2b9c6d849c97e304719140dcb967a9e2c254c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365780
Auto-Submit: Peter Kasting <pkasting@chromium.org>
Commit-Queue: Peter Kasting <pkasting@chromium.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43251}
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}
This is a reland of commit 65ae3245f9380e46b1d755f3f452ba63ab6cdf8d
with more backward compat which also fixes the off-by-one issue which caused wrong SRTP keys to be extracted.
Original change's description:
> Spanify SRTP key export
>
> and simplify the interface used as this is only used for exporting
> SRTP keys and passing arcane OpenSSL arguments around does not make
> much sense.
>
> BUG=webrtc:357776213
>
> Change-Id: I9e5a94fe368b77975e48b6dd5ab6a2d2575d6382
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364521
> Commit-Queue: Philipp Hancke <phancke@meta.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43198}
Bug: webrtc:357776213
Change-Id: I5d43dc23f90ef630834fb400751979fcc5e18203
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365180
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43225}
