since it contains helpers mostly related to cryptographically secure random numbers and strings.
BUG=webrtc:339300437
Change-Id: I10db939534b25dc792ac1600a4721d1b84521880
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352620
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42441}
Allow skipping the deinterleaving steps in PushResampler
before resampling when deinterleaved buffers already exist.
Bug: chromium:335805780
Change-Id: I2080ce2624636cb743beef78f6f08887db01120f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352202
Reviewed-by: Per Åhgren <peah@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42438}
The encoders stored in SimulcastEncoderAdapter might have a reference
which is passed by the encoder factory. Therefore, the encoder factory
needs to outlive the encoders.
A class inheriting SimulcastEncoderAdapter possibly owns the resource.
Since a base class' destructor cannot be called before a child class,
the child class needs to destroy encoders by invoking
DestroyStoredEncoders() in the destructor.
Bug: b:320555128
Test: Build
Test: unit test
Change-Id: I34672556bab5f6b0f754edb840e9eb42d38f292e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351442
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Hirokazu Honda <hiroh@google.com>
Cr-Commit-Position: refs/heads/main@{#42436}
The new version of MSan (rolled by [1]) detects the following:
```
==39908==WARNING: MemorySanitizer: use-of-uninitialized-value
#0 0x5591400a52ef in GetPlayoutDelayMs ./../../modules/audio_coding/neteq/decision_logic.cc:466:35
#1 0x5591400a52ef in webrtc::DecisionLogic::ExpectedPacketAvailable(webrtc::NetEqController::NetEqStatus) ./../../modules/audio_coding/neteq/decision_logic.cc:311:36
#2 0x5591400a39e9 in webrtc::DecisionLogic::GetDecision(webrtc::NetEqController::NetEqStatus const&, bool*) ./../../modules/audio_coding/neteq/decision_logic.cc:0:0
#3 0x55913cf590c9 in webrtc::DecisionLogicTest_PreemptiveExpand_Test::TestBody() ./../../modules/audio_coding/neteq/decision_logic_unittest.cc:139:3
#4 0x55913ef28283 in HandleExceptionsInMethodIfSupported<testing::Test, void> ./../../third_party/googletest/src/googletest/src/gtest.cc:0:3
#5 0x55913ef28283 in testing::Test::Run() ./../../third_party/googletest/src/googletest/src/gtest.cc:2710:5
#6 0x55913ef2ab46 in testing::TestInfo::Run() ./../../third_party/googletest/src/googletest/src/gtest.cc:2856:11
#7 0x55913ef2da34 in testing::TestSuite::Run() ./../../third_party/googletest/src/googletest/src/gtest.cc:3034:30
#8 0x55913ef621e8 in testing::internal::UnitTestImpl::RunAllTests() ./../../third_party/googletest/src/googletest/src/gtest.cc:5964:44
#9 0x55913ef60f54 in HandleExceptionsInMethodIfSupported<testing::internal::UnitTestImpl, bool> ./../../third_party/googletest/src/googletest/src/gtest.cc:0:0
#10 0x55913ef60f54 in testing::UnitTest::Run() ./../../third_party/googletest/src/googletest/src/gtest.cc:5543:10
#11 0x55913ee1a944 in RUN_ALL_TESTS ./../../third_party/googletest/src/googletest/include/gtest/gtest.h:2334:73
#12 0x55913ee1a944 in webrtc::(anonymous namespace)::TestMainImpl::Run(int, char**) ./../../test/test_main_lib.cc:203:21
#13 0x55913cbd36b8 in main ./../../test/test_main.cc:72:16
#14 0x7fdb18c73082 in __libc_start_main /build/glibc-LcI20x/glibc-2.31/csu/../csu/libc-start.c:308:16
#15 0x55913cb3e1a9 in _start ??:0:0
```
[1] - https://webrtc-review.googlesource.com/c/src/+/353620
Bug: b/344970813
Change-Id: I9b5d7791e68b4c494168ba9f007a3099ae21fed4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353581
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42433}
- avoid holding a lock across OnCaptureResult() callback to avoid a risk
of a possible deadlock
- annotate damage region as guarded by the same lock as latest frame as
both belong together
- document the acqusition order between locks
Bug: chromium:333945842
Change-Id: I9c65beed720ba54e40b85fb243a07d40524695f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353600
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Andreas Pehrson <apehrson@mozilla.com>
Cr-Commit-Position: refs/heads/main@{#42432}
The file base/allocator/partition_allocator/partition_alloc.gni reads
the variable is_cronet_build, which is defined by the cronet.gni file.
Removing the import and relying on the transitive import from another
.gni file, makes it brittle (e.g. in this case it breaks WebRTC's bots).
Bug: b/344727254
Change-Id: If6c537e934ae4b2243eb104a4b867b49f5d10cca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353540
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42431}
and files that broke when I fixed the first set.
Bug: webrtc:42226242
Change-Id: I321cd63537ab3002098c7bdecd889a6fc5a1eb25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353421
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42429}
Some `deps` entries may not be compatible with every system
that parses DEPS. This condition will be used to allowlist
deps slowly that are expected to work with every supported DEPS parser.
Bug: b/344601229
Change-Id: Idb076250f2ca4924603029e5206bcc6bc6c75231
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353440
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42426}
Mac OS X 10.5 was shipped in 2006, and Mac OS X 10.7 was shipped in
2010. Assume that WebRTC is not running on releases older than
those.
Bug: none
Change-Id: Ia7323c2ae7f186602aa972f390ea682bd2d1ff47
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353240
Auto-Submit: Avi Drissman <avi@chromium.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42423}
This is in preparation for moving the assignment of PT numbers
to a different place.
Bug: webrtc:42226302
Change-Id: I821f2b6b15eb9f9f3715714e2c2c220c5e5a219e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352921
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42422}
The elements in it can be packet better. This saves one byte.
The state cookie is only used from one peer to itself, so there is no
considerations around backwards or forwards compatibility.
Bug: None
Change-Id: I4f9a44f4c825626c7ed84bff52ed3155f5025a69
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353142
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Auto-Submit: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42421}
buildtools has first party support to fetch clang-format so no longer
needed to use custom hooks.
Bug: chromium:336843583
Change-Id: Ifceb292d05793496e1a3b5454e29e058c88ae205
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353040
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Josip Sokcevic <sokcevic@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42419}
PushResampler now uses a single buffer for the deinterleaved
channel sources and another for the deinterleaved destinations.
Before, there was a dedicated buffer per channel (source and dest).
This reduces allocations and allows for using DeinterleavedView for
both which simplifies some checks.
Bug: chromium:335805780
Change-Id: I553a36164109127fa332ab17918d53832d442303
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351542
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42415}
With this cl, sending can be forced with field trial "WebRTC-RFC8888CongestionControlFeedback/force_send:true/"
In the future, ReceiveSideCongestionController::EnablSendCongestionControlFeedbackAccordingToRfc8888 if RFC 8888 has been negotiated.
Bug: webrtc:42225697
Change-Id: Ib09066aa89ca7b3fffc551da541090c69ab8d75f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352720
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42413}
Interleave and Deinterleave now accept two parameters, one for the
interleaved buffer and another for the deinterleaved one.
The previous versions of the functions still need to exist for test
code that uses ChannelBuffer.
Bug: chromium:335805780
Change-Id: I20371ab6408766d21e6901e6a04000afa05b3553
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351664
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42412}
Without this, packets may be sorted in the wrong order.
Bug: webrtc:42225697
Change-Id: Ib9a72cdc7cb8f7ef6ca1571d095a6474215a83f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352821
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42411}
Before this change, the heartbeat timer was restarted every time a
packet was sent on the socket. On an idle connection, if the peer is
sending heartbeats, just responding to those heartbeats (with a
HEARTBEAT-ACK) would restart the timer, and then this socket wouldn't
do any heartbeating itself because the next hearbeat by the peer would
be received before the timer expires.
This is not according to the specification, where
https://datatracker.ietf.org/doc/html/rfc9260#section-8.3 states that
"A destination transport address is considered "idle" if no new chunk
that can be used for updating path RTT (usually including first
transmission DATA, INIT, COOKIE ECHO, or HEARTBEAT chunks, etc.)"
There are already timers running when INIT, and COOKIE-ECHO are sent
and not acked, so the heartbeat shouldn't be sent then. This is further
confirmed in the same section in the RFC which says that "The sending of
HEARTBEAT chunks MAY begin upon reaching the ESTABLISHED state". And
when INIT and COOKIE-ECHO are sent, the connection is not yet
established.
This CL changes so that the heartbeat timer is only restarted when any
DATA or I-DATA chunk is sent. This will make both sides send heartbeats
on an idle connection.
Bug: webrtc:343600379
Change-Id: I5ab159b7901e2ec9d37b24aaf845891b60a53c13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352841
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42409}
Because it is flaky !?
Bug: webrtc:42225697, b/343600373
Change-Id: I74415a9b97e90c25807b55053fd549f335b863ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352820
Reviewed-by: Markus Handell <handellm@google.com>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42408}
CongestionControlFeedbackGenerator collect receive time information about received
packets and sends feedback according to RFC8888
Bug: webrtc:42225697
Change-Id: I70b7f7322fd262f99f45fd56b6eb8630a11b30c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351543
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42404}
BWE logging has as far as I know know been used for a long time. RTC event logs are the prefered method of logging.
Removed since it causes some BUILD pain.
For debugging the metrics API https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/test/metrics/ can be used instead.
Bug: webrtc:343347276
Change-Id: I046b58d880faabfadbc22269b0392fdd644155fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352602
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42402}
This is a minor change for places where we use
AudioFrame::kMaxDataSizeSamples sized intermediary buffers. The change
uses `std::array<>` instead of C style arrays which allows for use
of utility templates that incorporate type based buffer size checking.
Also adding `ClearSamples()` method, which complements CopySamples.
Bug: chromium:335805780
Change-Id: I813feb32937e020ceb9ca4b00632dc90907c93fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351681
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42400}
To make it available for creating AudioEncoders in follow ups
Bug: webrtc:343086059
Change-Id: I24bb8f7e0494e392210cb1001ea0421030d2766b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352601
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42396}
Both CaptureFrame() and ProcessBuffer() hold a lock over the frame queue
and it happens that one waits for the other, causing unnecessary delays
since we already work with a queue having two frames, but this way we
don't really need a queue. Instead, keep reference to the last processed
buffer, which we will always use in CaptureFrame() and update it every
time at the end of ProcessBuffer(). This avoid unnecessary waiting for
the lock over the queue to be released.
Bug: chromium:333945842
Change-Id: I4afeb1daacd342e92578a50ac6e1c89a691bb8f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350042
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#42394}
Forwarding headers like rtc_event_log2_proto_include.h and test/gtest.h
were omitted.
Presubmit gn checks for existing (implicit) dependencies were disabled.
Bug: webrtc:42226242
Change-Id: Id08ae1b244db1a6f65069775f47deec05191ff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350923
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42393}
