This change adds support to allow ChromeOS capturers to also pass a
WindowId with a source. This WindowID can be used to help allow plumbing
and passing an Id that the capturing process knows about, in case it
wants to use any in-process capturing logic.
Bug: chromium:1273189
Change-Id: Ibcf494a75aec06eb1c44e6ff5fbdd9e2952e9b7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267086
Auto-Submit: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#38238}
This cl/ is a NOP refactoring,
moving the EncoderStreamFactory from within webrtc_video_engine.cc
into own file in video/. simulcast.cc is collateral.
Bug: webrtc:14451
Change-Id: Ia69b9241d8cd8a12be6628d887701f2e244c07cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276861
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38224}
This CL wraps the |Dav1dPicture| data directly for |VideoFrame| using
instead of copy data out to new buffer.
Bug: None
Change-Id: I21ceffb5cac7dda4a44eafbd0ed221974b8d45ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276526
Commit-Queue: Zhaoliang Ma <zhaoliang.ma@intel.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38194}
Shared screencast stream is tied to desktop capturer options,
which may outlive capturer itself. This leads to a case where
one may attempt to restart the stream in the capturer. This
causes the previous pipewire objects to leak (as observed
in `pw-top` output) and seems to appear as frozen screen for
clients. This CL ensures that the shared screen cast stream,
which is started in this capturer, is also stopped when the
capturer is destroyed.
Bug: chromium:1291247
Change-Id: I5f2b22e54e916549a5280ec457cd76360e42e48a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276640
Commit-Queue: Salman Malik <salmanmalik@chromium.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#38187}
Chrome Remote Desktop will support both X11 and Wayland desktop
capturers in the near future and we'd like to differentiate between
the two in our video frame stats and telemetry. I beleive other
products are in a similar position so I would like to add a capturer
ID to the frames generated by the capturer classes.
Bug: chromium:1366062
Change-Id: If27c35ad6ef89b6396120982edc4dd0cf2a1e51c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276081
Commit-Queue: Joe Downing <joedow@google.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#38185}
Reporting timeouts is useful for native hw backed codec implementations.
The value is in sync with VideoCodecStatus.java in the Android sdk.
Bug: b/185740707
Change-Id: I9a08a1303586c677be53aaa4f39455f42e519996
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276042
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Auto-Submit: Linus Nilsson <lnilsson@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38168}
Move functionality to closer where the values are used instead,
as per previous CL comment.
Bug: webrtc:14151
Change-Id: I6b7ca02da197420a1f5da930ba87021e6f557444
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275204
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38148}
- Propagating `RtpPacketInfo::local_capture_clock_offset`, an
existing field that is related to the abs-capture-timestamp
header extension field `estimated_capture_clock_offset`
- Propagated through `SourceTracker::SourceEntry`
Bug: webrtc:10739, b/246753278
Change-Id: I21d9841e4f3a35da5f8d7b31582898309421d524
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275241
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38129}
New ctor added without optional and media specific fields.
Bug: webrtc:10739, b/246753278
Change-Id: I7e15849aced6ed0a7ada725ea171a15ea1e9bc5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275941
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38124}
This add field trial string "skip_if_est_larger_than_fraction_of_max"
Dont send a probe if min(estimate, network state estimate) is larger than this
fraction of the set max bitrate.
Bug: webrtc:14392
Change-Id: I7333f6ef45ab0c019f21b9e4c604352219e1d025
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275940
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38123}
The reason for rejecting the congested frames in the first place, is
that they do not obey a Gaussian distribution around the line from the
Kalman filter. It therefore also does not make sense to include them
in the noise (*) estimation.
(*) noise = variation around the line from the Kalman filter.
Bug: webrtc:14151
Change-Id: Id8a44ba5f13bf9787ab54848109430ef7657f67a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275762
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38100}
This reverts commit a22c2a0c581cbe3f612f7a7d9fb9840186cc1e06.
Reason for revert: breaks upstream project
Original change's description:
> rtp sender: don't send BYE on deactivating streams
>
> as this breaks RTCP assumptions about SSRCs being no longer
> active as defined in
> https://www.rfc-editor.org/rfc/rfc3550#section-6.6
>
> This should not be sent in reaction to temporarily disabling
> a stream via RTCRtpParameters.active as this does not mean that
> the participant is leaving the session as defined in
> https://www.rfc-editor.org/rfc/rfc3550#section-6.3.7
> and does not indicate end of participation as defined in
> https://www.rfc-editor.org/rfc/rfc3550#section-6.1
> which stipulates BYE should be the last packet sent from this SSRC.
>
> BUG=webrtc:11082
>
> Change-Id: Ia5144857f85303643146b0759184f0f3f50b66e4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273348
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#38059}
Bug: webrtc:11082
Change-Id: Iaaff0c0d7bb857fe9ce78ebcc716f3c6f1bc5c4a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275640
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38097}
Opus was recently updated in Chromium (https://crbug.com/1347531), resulting in these failing for a non-SSE build.
Bug: None
Change-Id: I6c4124192f98f9358e7cc2241aec16a5338e689a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274760
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Felicia Lim <flim@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38095}
The `TimestampExtrapolator` is only used by the `VCMTiming`
class, despite there being references to it from both
`modules/rtp_rtcp/BUILD.gn` and `modules/video_coding/BUILD.gn`.
Bug: webrtc:14111
Change-Id: If1a02a56a0c83b13d619ca08dc76c884fa829369
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275482
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38093}
The Chromium RTCVideoEncoder unfortunately doesn't set if the
result is at target quality, and the definition of the threshold
is buried in libvpx_vp8_encoder.h.
This change
* Updates VideoStreamEncoder to postprocess an incoming EncodedImage
by interpreting the incoming QP information instead.
* Updates the related VideoStreamEncoder test to simulate an encoder
producing images around the QP threshold.
* Updates the steady state VP8 screencast QP threshold to a central
include file.
* Moves this and previously existing EncodedImage post-processing to a
new method AugmentEncodedImage.
Bug: b/245029833
Change-Id: I69ae29ffe501e84f28908f7d9a8cfd066ba82b43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275380
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38091}
This functionality should have been added as part of the original CLs,
but was missed. The purpose of the validation is avoid catastrophic
failures due to misconfiguration (such as RTC_CHECK crashes).
The purpose is not to always provide practically reasonable values.
Bug: webrtc:14151
Change-Id: Icbddade865bd6a868f467a1df7055026935f36f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275560
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38090}
The value is today set to 200 which is too low for an audio packet to trigger sending probes.
For the initial probing, it would be good if audio packets, that may arrive before the first video frame can trigger sending a probe.
Also fix field trial parsing of required number of probes.
Bug: webrc:14392
Change-Id: I1f3cebcda38b71446e3602eef9cfa76de61a1ccf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275620
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38089}
https://crrev.com/c/3885576 removed the last downstream consumer of the
constructor which took a ScreenCastPortal::CaptureSourceType. Now that
it has been removed, that constructor definition can also be removed and
the CaptureSourceType enum can be made private. There's still benefit in
storing and using this internally as the enum, since it's values match
that of the underlying system API.
The previously anonymous-namespaced function |ToCaptureSourceType| had
to be converted to a private static method as part of this change, since
it would be unable to access the type otherwise.
Fixed: chromium:1359411
Change-Id: I81ff24fbdddf9db02c9c5152d007dd82c194865a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274680
Auto-Submit: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#38084}
* Make `percentile` configurable and rename class.
* Introduce convenience type `MovingMedianFilter` that
maintains the behaviour of the old class with that name.
* Move home grown moving 95th percentile filter in
`JitterEstimator` to this new utility class.
Bug: webrtc:14151
Change-Id: I17d525b6e0bc98c28568c7dfe94b72eeab4a1ca2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275311
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38082}
kTemporalStructureMode4 is not used anywhere in the code.
Bug: None
Change-Id: I9a396f6706d26940fae68d1318942b5f31afa3bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274380
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#38079}
This change adds a median filter that can replace the
IIR filter that is currently used to estimate the
avg frame size (in bytes). It is enabled through a boolean,
and reuses the window length from the max percentile filter.
The median filter is only used by the delay calculation in
`CalculateEstimate()`. It does not replaced the use of the
IIR estimate in the size outlier rejection heuristic.
Bug: webrtc:14151
Change-Id: I519b6b57a8bee3c41a300ed2e92a1981c61cca15
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275121
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38077}
Most of the changes are trivial.
Bug: webrtc:14432
Change-Id: I0444527bf57c72c8d65f69754b4a4a1c1d7b2e92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275340
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38074}
Loss based BWE v2 rate is updated immediately when transport feedback is received.
This ensure that when GoogCcNetworkController::MaybeTriggerOnNetworkChanged is invoked, the loss based estimate is updated.
Bug: webrtc:14392
Change-Id: If404576c5793a29096cea52884862807cde8b615
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275306
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38070}
Detected by the new UBSan configurations (crbug.com/1352721):
[ RUN ] TestDecodingState.FrameContinuity
../../modules/video_coding/session_info.cc:250:10: runtime error: null pointer passed as argument 1, which is declared to never be null
../../build/linux/debian_bullseye_amd64-sysroot/usr/include/string.h:44:28: note: nonnull attribute specified here
No-Try: True
Bug: None
Change-Id: Ib4fb20d948b41da1a35dacb8abe944eb24f806f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275200
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38063}
as this breaks RTCP assumptions about SSRCs being no longer
active as defined in
https://www.rfc-editor.org/rfc/rfc3550#section-6.6
This should not be sent in reaction to temporarily disabling
a stream via RTCRtpParameters.active as this does not mean that
the participant is leaving the session as defined in
https://www.rfc-editor.org/rfc/rfc3550#section-6.3.7
and does not indicate end of participation as defined in
https://www.rfc-editor.org/rfc/rfc3550#section-6.1
which stipulates BYE should be the last packet sent from this SSRC.
BUG=webrtc:11082
Change-Id: Ia5144857f85303643146b0759184f0f3f50b66e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273348
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38059}
Main changes:
- `AgcManagerDirectTestHelper::FirstProcess()` replaced by
`CallAgcSequence()`, which is API contract compliant
- `ExpectCheckVolumeAndReset()`, `SetVolumeAndProcess()` and
`ExpectInitialize() `removed
- TODOs added for the next batch of improvements
- `AgcManagerDirectTestHelper::mock_agc` now using `NiceMock`
- `AgcManagerDirect::(AnalyzePre)Process()` now receives a
const ref
- `AnalyzePreProcess(const float* const*,size_t )` removed
Bug: webrtc:7494
Change-Id: Ie5bbaa590586dd806b30494fb00ca9c742c241e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273490
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38056}
The current design of the modified getter is error-prone since the
returned value changes meaning based on when (which point in the code)
the getter is called - namely, before `ProcessStream()` is called the
getter returns the stream analog level, after it returns the
recommended level.
Plus, the new implementation, which essentially returns a local
member, removes the risks that the non-trivial implementation
is computationally expensive.
Bug: webrtc:7494, b/241923537
Change-Id: I6714444df27bcc055ae693974ecd1f77f79e6ec0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271580
Reviewed-by: Hanna Silen <silen@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38055}
The `recommended_stream_analog_level()` getter is used to retrieve
both the applied and the recommended input volume. This behavior is
error-prone since the caller must know what is returned based on
the point in the code (namely, before/after the AGC has changed
the last applied input volume into a recommended level).
This CL is a first step to make clarity on which input volume is
handled in different parts of APM. Next in the pipeline: make
`recommended_stream_analog_level()` a trivial getter that always
returns the recommended level.
Main changes:
- When `recommended_stream_analog_level()` is called but
`set_stream_analog_level()` is not called, APM logs an error
and returns a fall-back volume (which should not be applied
since, when `set_stream_analog_level()` is not called, no
external input volume is expected to be present
- When APM is used without calling the `*_stream_analog_level()`
methods (e.g., when the caller does not provide any input volume),
the recorded AEC dumps won't store `Stream::applied_input_level`
Other changes:
- Removed `AudioProcessingImpl::capture_::prev_analog_mic_level`
- Removed redundant code in `GainController2` around detecting
input volume changes (already done by APM)
- Adapted the `audioproc_f` and `unpack_aecdump` tools
- Data dumps clean-up: the applied and the recommended input
volumes are now recorded in an AGC implementation agnostic way
Bug: webrtc:7494, b/241923537
Change-Id: I3cb4a731fd9f3dc19bf6ac679b7ed8c969ea283b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271544
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38054}
Switching to an AGC implementation agnostic solution for the input
volume emulation functionality offered by the
`capture_levels_adjuster` sub-module.
This CL also fixes a (silent) bug due to which, when the input
volume is emulated via the capture adjuster sub-module, AGC2
reads an incorrect value for the applied input volume.
Tested: audioproc_f with `--analog_mic_gain_emulation 1` used
to verify bit-exactness for one Wav file and one AEC dump for
which the input volume varies.
Bug: webrtc:7494, b/241923537
Change-Id: Ide3085f9a5dfd85888ad812ebd56faa175fb2ba7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273902
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38053}
`EchoControllerObservesAnalogAgc1EchoPathGainChange` is incorrect
since it does not call `set_stream_analog_level()`,
`ProcessCapture()` and `recommended_stream_analog_level()`
according to the contract.
`EchoControllerObservesNoDigitalAgc2EchoPathGainChange` is
useless since AGC2 doesn't have any analog controller at the
moment and the test is not written to explictly trigger digital
gain adaptations.
Bug: webrtc:7494, b/241923537
Change-Id: I56203c736448ec060077b00b57e98cd4c29fa737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271541
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38052}
Making it clear that the field is used to store the applied input
volume and not the recommended input volume.
Bug: webrtc:7494, b/241923537
Change-Id: Ib91bc1a12348f63e3a4ba6e068ed02e40786a87b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271342
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38051}
This CL adds the ability to forward aborted retransmission notifications
to specified RTP modules, as well as a way to find the RTX ssrc
associated with a media SSRC.
These will both be used by upcoming logic that can selectively flush
given streams from the pacer queue.
Bug: webrtc:11340
Change-Id: Ief3be47e4fd7dc5a1499bc21890e8979400ecb44
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274706
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38050}
This change adds a percentile filter that can replace the
"non-linear IIR" filter that is currently used to estimate the
max frame size (in bytes). The percentile filter is enabled through
the field trial, and it has two tuning parameters: the percentile
that is deemed the "max" frame, and the window length over which
the filter is applied.
Bug: webrtc:14151
Change-Id: I002609edb0a74161aaa6f0934892a1bec2ad8230
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274167
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38047}
