I've been working with SizeBench (binary size analysis tool) and it
reported that 39 virtual functions were not overridden. Removed
virtual keyword from each. SizeBench estimated waste 2.1kb. Change
made chrome.dll 5.3kb smaller. Since these 39 virtual functions
are never overridden, they are wasteful.
Note: These are the savings for Windows, relocation savings are probably larger on other platforms.
GN args for builds:
use_goma=true
is_debug=false
target_cpu="x64"
use_lld=false
fatal_linker_warnings=false
symbol_level=2
dcheck_always_on = false
pe_summarize analysis pre-change -> change:
Size of out\Default\chrome.dll is 187.205120 MB
Size of out\MediaContentDescription\chrome.dll is 187.199488 MB
Memory size change from out\Default\chrome.dll to
out\MediaContentDescription\chrome.dll
.text: -2624 bytes change
.rdata: -1984 bytes change
.pdata: -48 bytes change
.reloc: -644 bytes change
Total change: -5300 bytes
Bug: chromium:1371503
Change-Id: Ib33829fada54abdf8fed33ec96f11a03ce6fcb68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281442
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ivan Rosales <rosalesi@google.com>
Cr-Commit-Position: refs/heads/main@{#38630}
As the synchronous version only posts a task to recreate the encoder
later, it is not possible to catch errors and state changes that
could appear then.
The asynchronous version of SetParameters() aims to solve this by
providing a callback to wait for the completion of the encoder
reconfiguration, allowing any error to be propagate and subsequent
getParameters() call to have up to date information.
Bug: webrtc:11607
Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38627}
Motivation: loss based ramp-up can be incorrect when (1) bandwidth is loss limited, and (2) delay based estimate might be incorrect due to no delay signals. Therefore, bounding the loss based estimate by the delay based estimate is not much helpful in those cases.
Thus strengthening the bounding logic by using upper link capacity is one of solutions to avoid incorrect ramp-up.
Without the change: screen/qmLedxapJWvUTmn
With the change: screen/8sQcksWa6CptywK
Bug: webrtc:12707
Change-Id: I32ba82693b3ffa83cbb89c2cc9690fe16fb10c78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283085
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38626}
In the experiment WebRTC-SendPacketsOnWorkerThread ensure the safety
flag is set not alive even if Start/Stop has never been called.
Bug: webrtc:14502, chromium:1382602
Change-Id: I01c1e663762c8bb848e9bc31b2dcb22d38d0d1e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283380
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38624}
On Pixel 2, this causes an increase in flakiness. This needs to be
reenabled once the root cause is fixed.
Bug: chromium:1384172, b/259113795
Change-Id: Ie94d3e2daad3a2de5af673c763362ea1b42fde7d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283522
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38623}
Openh264 switched from api/svc to api/wels as the location for some
codec header files. During the transition it was necessary to
conditionally from either the old or new location, but now that the
switch is completed and has settled for about two weeks the conditionals
can be removed. This finishes the #include transition started by
webrtc-review.googlesource.com/c/280800
Bug: chromium:1218384
Change-Id: Ic0847428d134687908cc26fec1fdec0c612674b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Auto-Submit: Bruce Dawson <brucedawson@chromium.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38622}
BurstyPacer is currently controlled via field trials. In order for
Chrome to be able to have burst without relying on a field trial, this
parameter is added.
When all burst experiments have concluded we may be able to have a
hardcoded constant instead, but for now the parameter is added to
RTCConfiguration.
NOTRY=True
Bug: chromium:1354491
Change-Id: I386c1651dbbcbf309c15ea3d3380cf8f632b5429
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283420
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38621}
This is a reland of commit e6ec81a89ca904f1816b76456426babc28a9d767
Updated to ensure that the portal code can be built with is_chromeos.
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: I2c890c83c86ad60fa30f63dcf6fa90510d46009e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281661
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#38620}
Patchset 1 contrains the original cl.
Later patchsets contain fix.
Original description:
Continue probing if networkstat estimate increase
This fixes an issue where continues probing stops if networkstate estimate is low when a probe is sent, but increase as a consequence of the probe.
Bug: webrtc:14392
Change-Id: I8d4e1968020f9f8de18e12a4a0322a87f1a8fd2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283082
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38612}
As to not break downstream projects, the old name api/stats_types.h is
kept around to help include api/legacy_stats_types.h. We can delete this
in a follow-up.
NOTRY=True
Bug: webrtc:14180
Change-Id: I270ca5e366ae36e324cbc9f982bbb066ab92d203
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283081
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38610}
This reverts commit dd7dc25a30c5841e6620d195b83058a22ffff7cd.
Reason for revert: Bug in CL. Continuously probe if experiment for probing based on the link capacity is enabled.
Original change's description:
> Continue probing if networkstat estimate increase
>
> This fixes an issue where continues probing stops if networkstate estimate is low when a probe is sent, but increase as a consequence of the probe.
>
>
> Bug: webrtc:14392
> Change-Id: Id1d703f7efc824a6a6f8d899c367660291bd88c8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282941
> Reviewed-by: Diep Bui <diepbp@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38606}
Bug: webrtc:14392
Change-Id: Ib241b190951a78c436188c0b83d0247bf7d0dddd
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283080
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38609}
This is a reland of commit 937a59268e2ae56a58f648fba827444f7beb4466
Check if codec requested in createEncoder/Decoder is supported and return null if not.
Original change's description:
> Call native codec factories from Android ones.
>
> Android video codec factories are expected to be synchronised with the native ones in terms on supported codecs. But before this change there were differences:
>
> 1. Native decoder factory keeps AV1 support behind RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY while Android decoder factory advertises AV1 unconditionally;
>
> 2. Native encoder factory advertises AV1 if RTC_USE_LIBAOM_AV1_ENCODER is enabled while Android encoder factory never advertises AV1.
>
> This CL synchronises the codecs set in Android factories with that of native factories by calling native factories from Android ones.
>
> Bug: webrtc:13573, b/257272020
> Change-Id: I99d801eda0c5f3400bac222b9b08d719f1a6ed72
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282240
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38583}
Bug: webrtc:13573, b/257272020
Change-Id: Ida7bb9a2954b836a07ad560de29c1f8088264b77
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282802
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38607}
This fixes an issue where continues probing stops if networkstate estimate is low when a probe is sent, but increase as a consequence of the probe.
Bug: webrtc:14392
Change-Id: Id1d703f7efc824a6a6f8d899c367660291bd88c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282941
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38606}
This CL finalizes the Metronome refactor undertaken in
crbug.com/1381982 and enables it again in call.cc.
Fixed: chromium:1381982
Change-Id: I1642103e9c8a3f2a1f12d7635a1b27310802c1c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282920
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38605}
This reverts commit e87ec28b807f84babe228f54690c686fcf86a0fb.
Reason for revert: Breaks upstream.
Original change's description:
> Add checks for api/test mocks to make sure they're complete
>
> Also unifies the mock inheritance if they inherited from a ref counted
> interface:
> - it should only inherit from the interface
> - it should use make_ref_counted
>
> Bug: webrtc:14594
> Change-Id: I7b0514b632ccd0798028b50f19812ac0a196e13c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262423
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38602}
Bug: webrtc:14594
Change-Id: I9f2d9c3656b43e3006ec03ae7d792d0a53f47ebd
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282940
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38604}
Also unifies the mock inheritance if they inherited from a ref counted
interface:
- it should only inherit from the interface
- it should use make_ref_counted
Bug: webrtc:14594
Change-Id: I7b0514b632ccd0798028b50f19812ac0a196e13c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262423
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38602}
This is to avoid passing delay based estimate value twice from send side bwe.
Bug: webrtc:12707
Change-Id: Idc77cf7c2f4ecc60ae1dcfead325320532e7a7ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282864
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38600}
It appears to be still failing occasionally so add one more event
to verify streams connected successfully in order to verify whether
we sent and received buffers properly in the next step.
Bug: webrtc:14644
Change-Id: I08822b15452fc845d68cbff1b01ae6b6f7c1f486
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282842
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#38598}
Instead of disabling probing when the total allocated bitrate has
changed in goog_cc, it can be done via a new field trial parameter,
"probe_max_allocation". Not that the currently used flag
RateControlSettings::TriggerProbeOnMaxAllocatedBitrateChange() is per
default enabled and will be cleaned up in a follow up cl.
The field trial flag "skip_if_est_larger_than_fraction_of_max" now also
skip probing if the current estimate is larger than the currently max
allocated bitrate. ie, alr probing is skippe if the current estimate >
max configured bitrate or current estimate > max send bitrate of all
streams.
Bug: webrtc:14392
Change-Id: I2a09be39f85a9122410edd5acb1158ece12fca60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282860
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38597}
Add a few tests to get started on debugging.
The goal of this CL is to get the Fuchsia bots running the tests without infra specific issues. After landing this, failures will be in test framework files (e.g. test/testsupport folder) and WebRTC code.
Bug: b/232740856
Change-Id: I332607fe875334769e7dadf6696d878a23a7e69f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280440
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@google.com>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#38596}
Replace the use of MonoInputController::min_mic_level() with
MonoInputVolumeController::clipped_level_min() when estimating input
volume adjustment from clipping prediction. The adjustment is later
capped in MonoInputVolumeController::HandleClipping() using
clipped_level_min_ so no audio changes are expected from this change.
Bug: webrtc:7494
Change-Id: Ie26d0aa5cce3eeef06f70a281504889519bb5aca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282840
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38593}
Instead of trying to guess the state from the loss based estimator by
looking at the estimate, use the state.
Bug: webrtc:14392
Change-Id: Ibf6e762f02bfbfff175f2aa2bc98f45b1c5beb1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282823
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38589}
This is to ensure Epoch is the same if transport switch to TCP or another transport.
First packet received will always be timestamped with rtc::TimeMicros.
Other packet timstamps will use the kernel timestamp as an offset from the first packet timestamp.
For BWE, it is important that there is not a large time base diff if transport change.
This change is protected by the experiment WebRTC-SCM-Timestamp.
Bug: webrtc:14066
Change-Id: Iaeb49831e7019e21601bc90895ac56003a54e206
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281000
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38587}
Using ::recvmsg ensure packet timestamp can then be read directly when reading the buffer
instead of a separate system call and should also work on Ios/Mac.
The same experiment field trial flag will be "WebRTC-SCM-Timestamp/enabled/" and is also planned to be used for fixing webrtc:14066
Bug: webrtc:5773, webrtc:14066
Change-Id: I8a3749e87c686aa18fcee947472c1b602a0f63c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279280
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38585}
