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}
Tests currently rely on the sorted order of connections held within the ICE controller, which sorts the connections by usability. The internal ordering is not part of the ICE controller contract.
Tests use the ordering as a proxy for certain expectations, so changed the tests to explicitly test the expectations.
Bug: webrtc:14367, webrtc:1413
Change-Id: Iaf33c61f6eb968c2c93a0265b6c48ad6218e23a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275304
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#38088}
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}
This functionally no-op change adds the methods to allow an active ICE controller to manipulate the connection used by the ICE transport. Most methods reuse existing code, this will be explicitly marked for cleanup with a follow-up CL which adds active ICE controller support.
Non-trivial changes are needed for P2PTransportChannel unit tests to cover the new code, and these are also being added in a follow-up CL.
Bug: webrtc:14367, webrtc:14131
Change-Id: I4f012efcd8cb5766eb8c6f0872de50f8375f3a73
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275301
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#38081}
P2PTransportChannel currently relies on the ICE controller to keep track of this, even though P2PTransportChannel is actually supposed to hold the mutable connections.
Reading connections from the ICE controller also leaks some internal state from the ICE controller through the ordering of connections, which isn't strictly part of the interface. This change is a step towards fixing this.
This change is functionally no-op for now. The internal state will be used behind a field-trial in a future CL. That is also when some tests will be updated to work with the new internal state.
Bug: webrtc:14367, webrtc:1413
Change-Id: I6f8c5d805c780411fe940926f192fd2d6ce86d29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275081
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38080}
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}
P2PTransportChannel can then use either of the ICE controller factories configured with field trials.
Bug: webrtc:14367, webrtc:14131
Change-Id: I09ab99673d6ef81f56abe88987f5b67d84c24cb5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271292
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#38076}
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}
Plan B having been deleted from Chrome, there is no need to collect UMAs
relating to Plan B vs Unified Plan setups.
Bug: chromium:1357994
Change-Id: Icb5d16823ea9d849798583cd1c82683014b8a15c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275309
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38069}
This interface will be implemented by "new" ICE controllers that actively manage the connection used by the transport.
Bug: webrtc:14367, webrtc:14131
Change-Id: I0858884b0decd2a17ae9ca8617a043a085c61d54
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271291
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38066}
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}
This interface will be implemented by P2PTransportChannel in a follow-up CL. It will allow an ICE controller to request actions to manipulate the connection used by the transport.
Bug: webrtc:14367, webrtc:1413
Change-Id: I5cd171bd09c8dfc88588f8fc06e87d74a90b5216
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271290
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#38062}
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 increases the number of scenarios where the RTT would be
available to `ChannelReceive`. That's the case since
`ModuleRtpRtcpImpl2::RTT()` falls back on the DLRR-based method when
the report blocks based method is unavailable - i.e., when there is
no audio sender.
Bug: webrtc:10739
Change-Id: Ie2451c739ab5bcfbe7844ee852bb12a97dab2ca4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274581
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38048}
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}
