This should avoid some resampling before any packets have been received given that the vast majority of devices use 48k sample rate and the most common codec is Opus (which we always decode in 48k).
Bug: none
Change-Id: Ie7baea57c3eb1b763a6460c3b06b56d67b2b258e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280662
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38536}
Implement InputVolumeController and RecommendedInputVolumeEstimator based on the copy of agc classes AgcManagerDirect and MonoAgc.
Copies of the original files created in https://webrtc-review.googlesource.com/c/src/+/278624.
Bug: webrtc:7494
Change-Id: I74acee57b0db5cc8a6b666be9ba619c6c98a1773
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278625
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38533}
clarifying that the number of temporal layers is limited to
a single byte and moving the format description from the source
to the document.
drive-by editorial fixes
BUG=webrtc:12000
Change-Id: I33f85e0a81e1dc16ef762171c52a79919080e048
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279940
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38523}
This should be landed after https://chromium-review.googlesource.com/c/chromium/src/+/3986032
Bug: chromium:1218384
Change-Id: Id4104d2914f811e722a083021f515fd06b69b910
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280800
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Bruce Dawson <brucedawson@google.com>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38515}
The clang pragma have been added to ensure we can still test the code
until usage is gone, and that we can still have the one implementation
compiling without itself tripping on the deprecation errors.
Users of the code will have deprecation warnings or error as intended.
Bug: webrtc:14617
Change-Id: I21dae57c669557d4d218c235c811174a477be080
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281221
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38514}
Copy AgcManagerDirect files from agc to agc2. Rename the newly
created files and classes ahead of refactoring. Add a build
target.
This change is done to enable creating a class
InputVolumeController based on AgcManagerDirect. The added
temporary dependency on files in agc will be removed
in https://webrtc-review.googlesource.com/c/src/+/278625.
The exact copy of the files happened in the 1st patchset and it
has been verified as follows:
Checksum check:
```
$ git checkout main && git pull
# Go back to the tree state before [1] landed
$ git new-branch tmp
$ git reset --hard 2235776597e2f47ec353ac911428eb9a54d64a10
$ cd modules/audio_processing/agc/
$ md5 agc_manager_direct*
MD5 (agc_manager_direct.cc) = e661481a85f72596cae4599b62907f5b
MD5 (agc_manager_direct.h) = bf68280e2d0f689b4ebcd665b5db6052
MD5 (agc_manager_direct_unittest.cc) = 6bf0bf45ff5e940b1a3bb37154f09269
```
Patchset 1 (see [2])
```
$ cd modules/audio_processing/agc2/
$ md5 input_volume_controlle*
MD5 (input_volume_controller.cc) = e661481a85f72596cae4599b62907f5b
MD5 (input_volume_controller.h) = bf68280e2d0f689b4ebcd665b5db6052
MD5 (input_volume_controller_unittest.cc) = 6bf0bf45ff5e940b1a3bb37154f09269
```
[1] https://webrtc-review.googlesource.com/c/src/+/278781
[2] https://webrtc-review.googlesource.com/c/src/+/278624/1
Bug: webrtc:7494
Change-Id: I7804da899d18adf556b089c76a567ce27c299a62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278624
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38512}
This reverts commit e6ec81a89ca904f1816b76456426babc28a9d767.
Reason for revert: Assert on line 14, modules/portal/BUILD.gn breaks in downstream build. Reverting until it has been investigated.
Original change's description:
> Split out generic portal / pipewire code
>
> It will be reused by the video capture portal / pipewire backend.
>
> Bug: webrtc:13177
> Change-Id: Ia1a77f1c6e289149cd8a1d54b550754bf192e62e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263721
> Reviewed-by: Mark Foltz <mfoltz@chromium.org>
> Commit-Queue: Alexander Cooper <alcooper@chromium.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Salman Malik <salmanmalik@google.com>
> Cr-Commit-Position: refs/heads/main@{#38487}
Bug: webrtc:13177
Change-Id: I18deb5c78a54261f77693e7e31dba6f98f5eeb5d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280947
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Björn Terelius <terelius@webrtc.org>
Auto-Submit: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38496}
It will be reused by the video capture portal / pipewire backend.
Bug: webrtc:13177
Change-Id: Ia1a77f1c6e289149cd8a1d54b550754bf192e62e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263721
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Salman Malik <salmanmalik@google.com>
Cr-Commit-Position: refs/heads/main@{#38487}
This doesn't effect for how long the test will run, it just gives
PipeWire more time to establish connection and create empty buffers
before we try to work with it. All the waiting events will be
interrupted by signals once we no longer need to wait so it doesn't
matter if we wait 2 seconds or 5 seconds.
Bug: webrtc:14568
Change-Id: Ie918e8943bf882059b1289f57595fc302216745e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280700
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#38486}
This is a reland of commit 8d7273357d92fab881561d886ce8dfe94e6e2238
Root cause:
audioproc_f doesn't call `metrics::Enable()` and therefore the stats
reporter crashed when `metrics::HistogramFactoryGetCountsLinear()`
returned a nullptr.
Bug fix:
Added `InputVolumeStatsReporter::cannot_log_stats_`, a const flag
that is set to true if any histogram factory returns a nullptr.
When true, the class does nothing.
This CL also includes other code readability improvements that were
not part of the original CL.
Original change's description:
> APM: log both applied and recommended input volume stats
>
> This CL replaces the existing `WebRTC.Audio.ApmAnalogGain.*` stats
> with `WebRTC.Audio.Apm.AppliedInputVolume.*` and adds the
> `WebRTC.Audio.Apm.RecommendedInputVolume.*` stats.
>
> Bug: webrtc:7494
> Change-Id: I70be710d20b1589fc814cbce3d3329ac1500686f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280220
> Reviewed-by: Hanna Silen <silen@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38468}
Bug: webrtc:7494
Change-Id: I8373d16beb06b84f439d2c2274ededea7c5e95b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280661
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38484}
This metric was always supposed to be the spec's answer to
googBucketDelay, and is defined as "The total number of seconds that
packets have spent buffered locally before being transmitted onto the
network." But our implementation measured the time between capture and
send, including encode time. This is incorrect and yields a much larger
value than expected.
This CL updated the metric to do what the spec says. Implementation-wise
we measure the time between pushing and popping each packet from the
queue (in modules/pacing/prioritized_packet_queue.cc).
The spec says to increment the delay counter at the same time as we
increment the packet counter in order for the app to be able to do
"delta totalPacketSendDelay / delta packetSent". For this reason,
`total_packet_delay` is added to RtpPacketCounter. (Previously, the
two counters were incremented on different threads and observers.)
Running Google Meet on a good network, I could observe a 2-3 ms average
send delay per packet with this implementation compared to 20-30 ms
with the old implementation. See b/137014977#comment170 for comparison
with googBucketDelay which is a little bit different by design -
totalPacketSendDelay is clearly better than googBucketDelay.
Since none of this depend on the media kind, we can wire up this metric
for audio as well in a follow-up:
https://webrtc-review.googlesource.com/c/src/+/280523
Bug: webrtc:14593
Change-Id: If8fcd82fee74030d0923ee5df2c2aea2264600d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280443
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38480}
It doesn't really make sense to try to create the X11 capturer if we are
running under Wayland; nor does it make sense to create the PipeWire
capturer if we are going to fail to actually start a stream with it.
This change addresses both of these issues by exposing an IsSupported
method on BaseCapturerPipeWire and checking that we are not running
under Wayland before creating the X11 capturer.
Bug: chromium:1374436
Change-Id: Ieb291307376010e084824124ea8fde065545337c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279163
Auto-Submit: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#38474}
This CL replaces the existing `WebRTC.Audio.ApmAnalogGain.*` stats
with `WebRTC.Audio.Apm.AppliedInputVolume.*` and adds the
`WebRTC.Audio.Apm.RecommendedInputVolume.*` stats.
Bug: webrtc:7494
Change-Id: I70be710d20b1589fc814cbce3d3329ac1500686f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280220
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38468}
Adopt the new naming convention, which replaces "analog gain" and
"mic level" with "input volume", in the input volume stats reporter.
Bug: webrtc:7494
Change-Id: Ia24876151f51dd1dcc4e4f9db56c64d11ae3b442
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279840
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38467}
When packet arrives with large gap majority of the time could be spend
in finding next received packet. Embedding such search into PacketArrivalMap
makes it faster
Bug: chromium:1373414
Change-Id: I2e0be0f2fc4ea96af081531d575a17c70b72b25b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279881
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38459}
This new class implements the existing FieldTrialsView interface,
extending it with the verification functionality. For now, the
verification will only be performed if the rtc_strict_field_trials GN
arg is set.
Most classes extending FieldTrialsView today have been converted to
extend from FieldTrialsRegistry instead to automatically perform
verification.
Bug: webrtc:14154
Change-Id: I4819724cd66a04507e62fcc2bb1019187b6ba8c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276270
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38453}
Enable by default the look for the first echo.
Bug: webrtc:14205
Change-Id: Iae904679c1432f3a0766263907cf376903685b97
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278043
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38452}
Remove useless comments and properly test frame values. Also rename the
FakeScreenCastStream to TestScreenCastStreamProvider.
Bug: webrtc:13429
Change-Id: I9b1943f0903101a1d9228cded541d3766879d84f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279740
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#38450}
We already communicated SPA_META_VideoDamage before, but we never used
these metadata. This change checks whether SPA_META_VideoDamage metadata
are available and construct a damage rect combined from all sent damage
regions.
Bug: webrtc:13429
Change-Id: I326109b4bacf51855904e53345c671640d670323
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278820
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#38449}
This test created another PipeWire stream we can connect to with
SharedScreenCastStream and recieve frames from there. This is an
initial version, where I test whether we can successfuly connect
and disconnect, receive frames and it also tests DesktopFrameQueue.
In the future I will add tests to test mouse cursor and try to
come up with some corner cases and possible scenarios.
Bug: webrtc:13429
Change-Id: Ib2a749207085c6324ffe3d5cc8f2f9c631fa6459
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256267
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38431}
Tested: Chromium built with this change; verified that the
behavior at the beginning of the call has not changed with
both low (< 12) and high (> 12) input volumes.
Bug: webrtc:7494
Change-Id: Ie184c994d46bf6fd1cb209873383b911beb766e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278787
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38420}
RFC 3550 specifies samples to be the unit while https://w3c.github.io/webrtc-stats/#receivedrtpstats-dict* specifies time. This avoids the need to convert to time in code that reads the jitter value from RtpReceiveStats.
Bug: webrtc:13757
Change-Id: I972996971c58b686babd621ff4e0f5790fdf2cb1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279281
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Lundqvist <tomasl@google.com>
Cr-Commit-Position: refs/heads/main@{#38419}
The parameter is unused and it will be removed in [1]. This CL
isolates the necessary unit test changes from [1].
[1] https://webrtc-review.googlesource.com/c/src/+/278787
Bug: webrtc:7494
Change-Id: Ic1179d335926fba8ff1b65b494b538cf849724bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279100
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38416}
This reverts commit c1d5fda22c8ae456950c5549d22d099b478c67e2.
Reason for revert: This CL created thousands of metric alerts in the perf tests. It's possible that these are all expected, but since mbonadei@ is OOO right now, I think it's better to revert, and have him re-land when he is back.
Most alerts are here: https://bugs.chromium.org/p/webrtc/issues/detail?id=14549
Original change's description:
> Add documentation, tests and simplify webrtc::SimulatedNetwork.
>
> This CL increases the test coverage for webrtc::SimualtedNetwork, adds
> some more comments to the class and the interface it implements and
> simplify the logic around capacity and delay management in the
> simulated network.
>
> More CLs will follow to continue the refactoring but this is the
> ground work to make this more modular in the future.
>
> Bug: webrtc:14525, b/243202138
> Change-Id: Ib0408cf6e2c1cdceb71f8bec3202d2960c5b4d3c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278042
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38388}
Bug: webrtc:14525, b/243202138
Change-Id: I5bc56c954bb12e7c27cb859e838f0b7a89e006f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279522
Owners-Override: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38415}
This reverts commit 332810ab5d41862b8f85ef30e84dbec4241f8b21.
Reason for revert: This commit chain seems to cause problems in LossBasedBwe.
Original change's description:
> Probing integration in loss based bwe 2.
>
> - Loss based bwe has 3 states: increasing (increasing when loss limited), decreasing (decreasing when loss limited), or delay based bwe (the same as delay based estimate).
> - When bandwidth is loss limited and decreasing, and probe result is available, GetLossBasedResult = min(estimate, probe result).
> - When bandwidth is loss limited and increasing, and the estimate is bounded by acked bitrate * a factor.
> - When bandwidth is loss limited and probe result is available, use probe bitrate as the current estimate, and reset probe bitrate.
>
> Bug: webrtc:12707
> Change-Id: I53cb82aa16397941c0cfaf1035116f775bdce72b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277400
> Commit-Queue: Diep Bui <diepbp@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38382}
Bug: webrtc:12707
Change-Id: Ied86323b0ce94b87ac503a2ee34753cebef5f53d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279500
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38412}
This reverts commit aa71259b06b72ba0fc5a28c0ffa4891c69c09441.
Reason for revert: This commit chain seems to cause problems in LossBasedBwe.
Original change's description:
> Probe when network is loss limited.
>
> Trigger probes next process intervals if the loss based current state is either increasing or decreasing. 0/ first probe at the loss based estimate. 1/ if increasing: allow further probing. 2/ if decreasing: not allow further probing.
>
>
> Bug: webrtc:12707
> Change-Id: I4e99edcbe4e2c315e8498ffb7fb2e589cdb4e666
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279041
> Commit-Queue: Diep Bui <diepbp@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38395}
Bug: webrtc:12707
Change-Id: I1fb61337148faf6faaa0056dc25f14536a19a462
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279480
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38410}
Trigger probes next process intervals if the loss based current state is either increasing or decreasing. 0/ first probe at the loss based estimate. 1/ if increasing: allow further probing. 2/ if decreasing: not allow further probing.
Bug: webrtc:12707
Change-Id: I4e99edcbe4e2c315e8498ffb7fb2e589cdb4e666
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279041
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38395}
Libdrm is an essential library and should be available everywhere where needed. It also looks it's a dependency for Chromium already.
Bug: webrtc:13429
Change-Id: Id81497b4f29bbd80f7d94f57333aa533288c3538
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279023
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#38392}
This CL increases the test coverage for webrtc::SimualtedNetwork, adds
some more comments to the class and the interface it implements and
simplify the logic around capacity and delay management in the
simulated network.
More CLs will follow to continue the refactoring but this is the
ground work to make this more modular in the future.
Bug: webrtc:14525, b/243202138
Change-Id: Ib0408cf6e2c1cdceb71f8bec3202d2960c5b4d3c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278042
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38388}
- Loss based bwe has 3 states: increasing (increasing when loss limited), decreasing (decreasing when loss limited), or delay based bwe (the same as delay based estimate).
- When bandwidth is loss limited and decreasing, and probe result is available, GetLossBasedResult = min(estimate, probe result).
- When bandwidth is loss limited and increasing, and the estimate is bounded by acked bitrate * a factor.
- When bandwidth is loss limited and probe result is available, use probe bitrate as the current estimate, and reset probe bitrate.
Bug: webrtc:12707
Change-Id: I53cb82aa16397941c0cfaf1035116f775bdce72b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277400
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38382}
This CL:
- makes it easier to understand the (nontrivial) metric interpretation
- corrects the computation of BufferDelay to use 0 for absent delay
- deletes metric MaxSkewShiftCount, unused since https://webrtc-review.googlesource.com/c/src/+/119701
- updates the unit test to directly test metric reporting
Corresponding update to histograms.xml:
https://crrev.com/c/3944909
Previous revert:
https://webrtc-review.googlesource.com/c/src/+/279040
This CL is identical to the original, except:
- the test is updated to spam fewer EXPECT_EQ failures on failure (EXPECT_EQs moved out of inner loop)
- the test not resets metrics (metrics::Reset()) at the beginning, like other histogram tests
Bug: webrtc:8671, chromium:1349051
Change-Id: Ie802e1f9d03a22ff7018f522a63b19e0b6eec2e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279046
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38376}
This is a speculative fix for the DCHECK at the top of
ScreenCapturerX11::CaptureScreen(). Whenever |selected_monitor_rect_|
changes, |queue_| should be reset, so that new frames are allocated
with the correct size. This CL adds a reset to UpdateMonitors() which
modifies |selected_monitor_rect_| and is called whenever an X11
configuration-change event is received (for example, when a monitor is
resized).
Bug: chromium:1372579
Change-Id: I9cc84a8b6990802f9d7dde05966ee17a80ddd48e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279065
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Lambros Lambrou <lambroslambrou@chromium.org>
Auto-Submit: Lambros Lambrou <lambroslambrou@chromium.org>
Cr-Commit-Position: refs/heads/main@{#38374}
