For now use doubles as units in api/units have insufficient precision for jitter estimation.
Bug: webrtc:14381
Change-Id: I5a691b6a404b734a5bef11d677b72040bc02ff0f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272367
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37841}
Now Configure(), Decode() and Release() calls to the decoders should
all happen on the decoder thread. Added thread checkers to verify.
Bug: None
Change-Id: I2a1cf2cf7f3c3c7c50e382d82a3638e916ed9c34
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272368
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37840}
Re-enabled multithreaded encoding using OpenH264, as the issue described in crbug.com/583348 no longer applies.
Bug: webrtc:14368
Change-Id: I5ae768a6edf3b40d99c13fb4ee4662626c993a66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271820
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37837}
As mentioned in https://crbug.com/webrtc/11956, the results did not show
any performance improvments.
Bug: webrtc:11956
Change-Id: Ie050aa5a6083fcf0c776fb8d03e7d18644b37f97
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272280
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37833}
This was only set to nullptr in non-test environments and was thusly
unused. With this change, the stats callbacks are gaurenteed to only
come from the VideoStreamBufferController and so the thread checks can
be removed.
Bug: webrtc:14003
Change-Id: Iaf0e77aa7c45a317e38ae27739edcefd3123d832
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272021
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37816}
I've added the proper headers to the only file in Chromium which includes screen_capture_frame_queue.h (see https://chromium-review.googlesource.com/c/chromium/src/+/3836317).
I've also built the remoting host and Chrome on Windows and Linux with this change and did not see any build errors.
The only build error I encountered was in shared_screencast_stream when building webrtc so I added the required header there.
Bug: webrtc:14378
Change-Id: Ie88e606dfa52f18514a87b87e5904424543d7df3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271922
Commit-Queue: Joe Downing <joedow@google.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37811}
Moves FrameBuffer2 to its own GN target to reduce the binary size of the
video target.
Bug: None
Change-Id: I40e86a1eabc0c9e8e6fada3dcdb4e3a043c61c6c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271286
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37803}
- Move existing check on `max_frame_size` to the top.
By doing this early, the filter will not end up in an
inconsistent state (predicted but not updated) when
called with a tiny `max_frame_size`.
- Add sanity check on noise variance.
This will avoid sqrt of a negative number.
Bug: webrtc:14151
Change-Id: I2507a9114655d3c3ae35bf5ed83f3f1154c42ad3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271281
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37798}
In migrating rtc::Event to use TimeDelta instead of int,
rtc::Event::kForever will have to become something else.
This change removes dependencies on that kForever is int.
Bug: webrtc:14366
Change-Id: Ic36057dda95513349e7ae60204e7271ff1f58825
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271288
Auto-Submit: Markus Handell <handellm@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37795}
This CL reorders the update steps physically, to make them
easier to comprehend. It renames variables to be more verbose,
but also adds succinct mathematical descriptions (using Wikipedia
notation) to all steps.
No functional changes are intended with this change.
Bug: webrtc:14151
Change-Id: I6a4642e89e2b73639f0b4c928e07b317c14d5884
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271546
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37784}
This class is never overridden.
Bug: webrtc:14151
Change-Id: I3b70e927ee0eafd71ce4bdb8f6e8d6330c1a3f08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271501
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37781}
* Update naming of data members.
* Start reordering code blocks in `PredictAndUpdate`.
(The "predict" step is done in this C:. The "update"
step will be improved in another cl.)
Bug: webrtc:14151
Change-Id: Idea1e8e786bd672dadedbcb3cd5598f4a033e81e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271023
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37767}
This CL simplifies and documents the interface of the Kalman filter better. A simple unit test verifying the filter's convergence is
added. No functional changes to the filter are intended.
Future CLs will improve the data member naming and code organization.
Bug: webrtc:14151
Change-Id: I01e60d4b783cabad3ccbdc711c5d746666dd16ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265877
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37766}
Instead of showing individual byte differences, this CL detects
differences in the expected and actual byte streams of the evaluated
AEC dump and, if detected, parses the `audioproc::Event` proto lite
messages and calls EXPECT_EQ() for a subset of individual (sub-)fields.
Note that messages are parsed only if the byte streams of each message
pair do not match, so with no failures the test runs at no extra cost.
Plus, the the added funcionality can only be enabled for local
debugging by flipping the `kDumpWhenExpectMessageEqFails` flag - a
code change cannot land if the flag is set to true.
Note that `MessageDifferencer` (see [1]) could not be used because
it is not implemented for `MessageLite` protos.
[1] https://developers.google.com/protocol-buffers/docs/reference/cpp/google.protobuf.util.message_differencer
Bug: b/241923537
Change-Id: I8e0eda3b1ecfe06945b6dad5ee8871f8200d76d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270922
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37765}
When ScreencastPortal::OnStartRequestResponseSignal receives either a
non-zero response code or is missing the response data, it would
directly cast this to a RequestResponse. However, this direct cast is an
error. Per the documentation, the response signal returns the following
values with their corresponding meanings:
0 - Success
1 - User Cancelled
2 - Error
The RequestResponse enum however, has "kUnknown" as 0, and thus
"kSuccess" as 1 (with all other values also shifted up by 1 value). This
means that when the portal was cancelled, we were still receiving
RequestResponse::kSuccess. This fixes the issue by removing the improper
cast and adding a translation function. This function is local for now
since no where else attempted to cast values to a RequestResponse; but
can be moved if the need arises.
Fixed: chromium:1351824
Change-Id: I4cd44d90055147c9592d590c7969dcfc3297a3d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271240
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Auto-Submit: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37755}
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}
