that rtc::Location parameter was used only as extra information for the
RTC_CHECKs directly in the function, thus call stack of the crash should
provide all the information about the caller.
Bug: webrtc:11318
Change-Id: Iec6dd2c5de547f3e1601647a614be7ce57a55734
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270920
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37748}
Remove check if `prev_estimate_` is less than 10 us since it can never
be less than 1 ms.
Bug: None
Change-Id: If151390d22fa0b6ecdc36af64168d3e2049c7b6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271203
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37745}
The data that's used to report the histograms is owned by UlpfecReceiver
and moving the reporting there, simplifies things as configuration
changes happen in RtpVideoStreamReceiver2 (which currently require all
receive streams to be deleted+reconstructed).
Additional updates:
* Consistently using `Clock` for timestamps. Before there was
a mix of Clock and rtc::TimeMillis.
* Update code to use Timestamp and TimeDelta.
Bug: none
Change-Id: I89ca28ec7067a49d6b357315ae733b04e7c5a2e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271027
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37729}
replace std::deque implementation with a manually controlled circular buffer.
replace Timestamp validity check from 'IsInfinite()' accesser to cheaper comparison to zero.
These greatly increase PacketArrivalTimeMap::AddPacket perfomance when packet arrive with large sequence number gaps.
Bug: chromium:1349880
Change-Id: I6f4e814b1086ca9d0b48608531e3a387d9e542dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270564
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37722}
The intention of this change is to separate the Kalman filter state
(that prior to this change lived in JitterEstimator) from the
other filter's state, making it easier to see how the different
filters interact.
This move does not include any interface, functional, or
documentation changes. Those will follow in later changes.
A very basic unit test is added, which will also be expanded
later on.
Bug: webrtc:14151
Change-Id: Ifb9b8ce2d9418ea52ccf64a77fd46d1ebba30779
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264984
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37721}
We pass the fd we recieve from xdg-desktop-portal to PipeWire to connect
to it and according to the specification PipeWire automatically closes
it on disconnect or failure. We also close the fd ourself when we tear
down the portal connection so we have to avoid doing this twice. Looks
OBS studio just duplicates the fd passed to PipeWire so do the same in
order to avoid the fd ownership violation once we stop sharing.
The fd we recieve from xdg-desktop-portal is from PipeWire also using
fcntl() with F_DUPFD_CLOEXEC option.
Bug: chromium:1339236
Change-Id: Ia7aee36e520dd5ff9a40688a6807e31c4e636f8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270421
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37712}
* Add test to Generic decoder unittests to ensure drop behaviour is covered.
* Use simulated time in the generic decoder unittests.
Bug: webrtc:14324
Change-Id: I10b28b45c434f92d5344683fb9ca6676efe0e08c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270662
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37710}
Sample rates not divisible by 100, in particular 11025 Hz and 22050 Hz, have long been used with APM in Chrome, but the support has never been stated explicitly.
This CL makes minor modifications to the APM API to clarify how rates are handled when 10 ms is not an integer number of samples. Unit tests are also extended to cover this case better.
This does not update all references to 10 ms and implicit floor(sample_rate/100) computations, but it does at least take us closer to a correct API.
Note that not all code needs to support these sample rates. For example, audio processing submodules only need to operate on the native APM rates 16000, 32000, 48000.
Bug: chromium:1332484
Change-Id: I1dad15468f6ccb9c0d4d09c5819fe87f8388d5b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268769
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37682}
follow-up from https://webrtc-review.googlesource.com/c/src/+/262810
* replace Time::Millis(0) and TimeDelta::Millis(0) with ::Zero()
* drop unnecessary webrtc namespace from some TimeDeltas
* make TimeDelta do the unit conversion for stats
BUG=webrtc:13756
Change-Id: Ic60625ae0fc7959a47a6be9f5051851feaf76373
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265875
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37664}
This should avoid any potential race in static initialization for the
tracing logic.
Bug: webrtc:12715
Change-Id: Ic91d8e5fbd9b45a91e7e7a9e76226fc558e00c4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270381
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37662}
Some of the timestamps input into UpdateCurrentDelay are not truncated
to milliseconds and thus a small negative delay can result. This means
the delay will not update when it should have.
Bug: webrtc:14168
Change-Id: I5293339b6a39201c680854e9596b717025ee8dc1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266370
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37657}
Before this change the full screen application handler was failing to
detect PowerPoint going into presentation mode, resulting in the editor
window continuing to be shared rather than the intended behavior of
sharing the presentation itself.
Fix this by always looking for the PowerPoint full screen presentation
window, regardless of whether the editor window is still open. In
the current version of PowerPoint, the editor stays open during
presentation.
Bug: chromium:1231437
Change-Id: I1b21e263d25320cc236d127d22d4d64bb52fcbda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269560
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37632}
This is the first step of migrating
AudioProcessing::CreateAndAttachAecDump() from using std::string to
absl::string_view.
Bug: webrtc:13579
Change-Id: I8fc373e7ac55fd8e96bb0b01d1a30e28883ac9a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269400
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37631}
This overload was removed in C++20.
Bug: chromium:1284275
Change-Id: I67a25ae23fa111e4972d1b207f1c078da13d86a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269440
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Auto-Submit: Peter Kasting <pkasting@chromium.org>
Commit-Queue: Peter Kasting <pkasting@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37630}
WebRTC doesn't produces such packet and ignores it when receive.
Bug: None
Change-Id: I4af8cb3308cb2422808bdfc420a85fa175085bfb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269181
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37627}
instead of using Lock/Unlock attributes, use Assert attribute to annotate code is running on certain task queue or thread.
Such check better matches what is checked, in particular allows to
recheck (and thus better document) currently used task queue
Bug: None
Change-Id: I5bc1c397efbc8342cf7915093b578bb015c85651
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269381
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37619}
Replace helper functions with the constant
Remove option to set min bitrate in RemoteBitrateEstimator as unused:
ReceivedSideCongestionController is the only user of the
RemoteBitrateEstimator interface, and it always sets the same value
right after construction that RemoteBitreateEstimators already use.
Bug: None
Change-Id: If179fdd72b1ded6ad1fd0a6dfffc97b302153322
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269383
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37613}
Keeping the field trial around as a kill-switch for another milestone.
Bug: webrtc:11340
Change-Id: I3285baefab176f541cbb5ed3bacbc669d3e8836f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269384
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37611}
Inserting packet with zero arrival time may trigger inconsistent state in the internal map where packet sometimes treated as received, but sometimes treated as not received.
Bug: chromium:1346959
Change-Id: I0809e41a873103dcd62528358e64794c1d3cb28f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269382
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37609}
1. Add loss threshold for high bandwidth preference. If the average loss ratio is less than the threshold, then the model prefers higher bandwidth candidates. Otherwise, it prefers lower bandwidth candidates. Before, it always prefers higher bandwidth candidate. The default value is 0.99, means it always prefers high bandwidth candidates.
2. Only increase the estimate if the inherent loss (random loss) is equal to/greater than the average loss. If the inherent loss is less than the average loss, then it is oversending, thus should not increase the estimate.
Bug: webrtc:12707
Change-Id: I37eb536679ca29e017a4a47703b417efd4d103ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269101
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37608}
Consumers expect the frame to be valid if Result::SUCCESS is delivered.
If the frame is nullptr, we should deliver ERROR_TEMPORARY instead.
Bug: webrtc:14265
Change-Id: If94a3ead38d7657d7b90bbe046256be697312216
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269223
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37590}
Update an affected unit test by the change in goog_cc.
Bug: webrtc:14272
Change-Id: I83e97530c861b126bed876d57f6d4f91aa45de7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269002
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37587}
It is now easier to fully test `AgcManagerDirect` with different values
for the used field trials. In particular, this CL adds tests for the
field trial named `WebRTC-Audio-2ndAgcMinMicLevelExperiment`.
1. `UnmutingRaisesTooLowVolume` and `MicVolumeIsLimited`
The expectations for the lowest input volume are not hard-coded anymore
since the parametrized tests use different values for the enforced
minimum.
2. `RecoveryAfterManualLevelChangeBelowMin`
The recovery behavior after manual input volume change depends on
whether the minimum input volume is overridden. When that's the case,
the minimum volume is applied immediately after the manual adjustment.
Hence, the existing test is left and a parametrized version of it has been added to test the "instant recovery" behavior. The latter test is
skipped when the minimum input volume is not overridden since that case
is covered by the existing test.
Bug: chromium:1275566
Change-Id: Ib0d4427b32b88f33138d4062b365916a3c47a406
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268900
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37577}
Using 4 temporal layers is not quite supported: Not advertised, no
integration tests. When transitioning to configuration via scalability
mode, there are no corresponding modes defined. So delete these two
tests; they can be added back if/when support for corresponding
scalability modes are added.
Bug: webrtc:11607
Change-Id: I97f55dc95d6513ccf65fa887757a62e9c8659be7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269003
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37575}
Stop using TEST_F; that will make it easier to switch to parametric
tests that are needed to correctly test `AgcManagerDirect`.
"Avoid fixtures where reasonable."
Source: https://abseil.io/tips/122
Bug: chromium:1275566
Change-Id: I2d73a0913eb2349144f63bd17ab4d6efa245e472
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268766
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37556}
rtc::TaskQueue is a simple wrapper over TaskQueueBase and adds no
extra features when task queue is used without passing ownership.
Reducing usage of the internal rtc::TaskQueue wrapper gives users more flexibility how TaskQueueBase* is stored.
Bug: webrtc:14169
Change-Id: If5c8827544c843502c7dfcef775ac558de79ec3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268189
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37549}
