This is a partial revert of the previously landed CL in
https://webrtc-review.googlesource.com/c/src/+/301200.
I noticed when I used `getDisplayMedia` on my private Windows laptop
in combination with WGC that the captured screen was distorted and
did only contain tilted (~35 degrees) lines in all sorts of colors.
The issue only happened for one particular screen resolution
3000 x 2000 and 200% scale. Changing to 100% scale instead resolved
the issue.
I tried many other resolutions but could only trigger for the one
above with 200% scaling.
Next, I bisected and found [1] which led to [2] which contains my own
change in https://webrtc-review.googlesource.com/c/src/+/301200.
The only part that could affect the video frame was the part which
did `CopyPixelsFrom` so I reverted that part and it solved the issue
on my private Windows laptop.
I did not dig deeper into why this particular resolution triggered
the distortion but deiced to revert to avoid more reports.
[1] b4c2a7fcf5..ff848b7a43
[2] a9a2957dbc
Bug: chromium:1428592
Change-Id: I328e77840cd3ca6871254cdf06500bdc616b0c36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306600
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#40147}
Instead of using most recent, or most "valuable" packets for padding, use most recent large packet.
The large packet for padding is not culled when acked by the receiver.
The most recent large packet is kept where payload size + 100bytes > currently stored.
Bug: webrtc:15201
Change-Id: I510735b757f99460c477b575061963d2b69016e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306521
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@google.com>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40146}
Generalize the threshold schema to support arbitrary metrics.
This is part of migrating to the new schema.
The old bug_filing_threshold and priorities.threshold will be removed once the migration is done.
Bug: None
Change-Id: I7d79b2f43a6ce7ad187fd1e75663d0629723325f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306560
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Commit-Queue: Beining Chen <beining@google.com>
Cr-Commit-Position: refs/heads/main@{#40145}
This change allows us to remove one static_cast from tests that
was problematic for another refactoring.
Bug: webrtc:13931
Change-Id: I8e1b5cecadd806b266b6c115b56b18b9613cbe82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306500
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40144}
which is the correct term used in
https://www.rfc-editor.org/rfc/rfc3611#section-4.4
BUG=None
Change-Id: Iab5a1de6b69a8495aa9a6f79531053f4f2421c27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306480
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40143}
Whether a metric is to be exposed to JavaScript or not is a blink
implementation detail that the WebRTC repository does not need to be
concerned with.
This CL removes unused code and paves the way for the possibility of
making the one and only RTCStatsMember class be absl::optional<>-based
in the future.
Bug: webrtc:15162
Change-Id: I578715f48b8fcc3534b72b4c700fd6567f8d553e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304722
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40139}
Prior to this CL, the video `jitterBufferDelay` stat was the accumulated current delay, which is a smoothened version of the target delay. This is not correct according to the spec [1]. Rather, the stat should be the accumulated time spent in the jitter buffer, for all emitted frames. This CL fixes this spec compliance problem.
Expect changes to test metrics and product monitoring as this CL rolls out.
[1]: https://www.w3.org/TR/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbufferdelay
Tested:
1. Go to https://jsfiddle.net/jib1/0L6duga2/show
2. Apply 2.0 seconds of video delay.
3. Notice that "Video jitter buffer delay" is slightly less than 1990ms. (2000ms playoutdelayhint - 10ms render delay - Xms decode delay).
Bug: webrtc:15085
Change-Id: I42805faafd7dd3bcdcf3ad08e751e08d6de38906
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304521
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40138}
Replace Transport* interface with since std::function to stress this class doesn't produce RTP packets
Repesent outgoing packet as ArrayView instead of pointer + length.
Make outgoing transport optional, thus allowing to use RtcpTransciever as an rtcp parser.
Bug: webrtc:8239, webrtc:14870
Change-Id: Ia582d9a980786df8e295adcebe27081258b80dc0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306280
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40134}
in particular which Chromium directories are taken into account.
Also update the list of release notes.
BUG=webrtc:14227
No-Try: true
Change-Id: I7279e1e79cb397e154bc8c6545f323d194ab8f08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306360
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Christoffer Jansson <jansson@google.com>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40133}
NonStandardGroupId is no longer wired up to Chrome, but if we did want
to only expose certain metrics if a field trial was enabled then the
right place to do that would be in blink, where WebIDL lives.
This was only used prior to the WebRtcStatsReportIdl launch and
experiments haven't been active in several years so its dead code.
Blocked on:
- https://chromium-review.googlesource.com/c/chromium/src/+/4514754
Bug: webrtc:15162
Change-Id: Ia41a4d21d7e5f029ddb121183fbd69ae7f98fac4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304720
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40132}
This is a refactoring that should have no production impact.
It has been activated for 2 weeks before, but was rolled back
because of a performance impact - this has now been fixed.
Intended to be submitted May 24 - after the 115 branch cut.
Bug: webrtc:13931
Change-Id: I745558cc3062cb4ea0a4d6f537702efc96eb7574
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305221
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40131}
Track stats are roughly equal in size as the RTP stream stats which
are the largest objects making up the majority of the RTCStatsReport
size and scales with meeting size. Deleting track/stream reduces the
size in approximately half which should reduce performance overhead
and unblock code simplifications.
Blocked on:
- https://chromium-review.googlesource.com/c/chromium/src/+/4517530
# Relevant bots already passed
NOTRY=True
Bug: webrtc:14175, webrtc:14419
Change-Id: Ib7bdb84c10459b42b829228d11876498e5227312
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289043
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40129}
This reverts commit 8856410b6d54b546bdb3185587474f0f9b3a7c2e.
Reason for revert: chromium:1447540
Original change's description:
> pipewire capturer: Reduce the amount of copying
>
> Improves the capture latency by reducing the amount of
> copying needed from the frame. We keep track of the
> damaged region of previous frame and union it with
> the damaged region of this frame and only copy this
> union of the frame over. X11 capturer already has
> such synchronization in place.
>
> The change is beneficial especially when there are
> small changes on the screen (e.g. clock ticking).
> For a 4k screen with 128 cores, I observed the
> capture latencies drop from 5 - 8 ms to 0 ms when the
> system is left idle. This is in line with the X11
> capturer.
>
> Bug: chromium:1291247
> Change-Id: Iffb441f9e1902d2658031f5f35b5372ee8e94073
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299720
> Reviewed-by: Alexander Cooper <alcooper@chromium.org>
> Commit-Queue: Salman Malik <salmanmalik@chromium.org>
> Cr-Commit-Position: refs/heads/main@{#39968}
Bug: chromium:1291247
Change-Id: Id1bfd3fc39fea2bb1f232cad5218f90e144920e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306263
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Auto-Submit: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#40123}
EglThread can be shared by multiple clients each using their own
EglBase instance, but sharing thread and EglConnection.
go/meet-android-eglcontext-reduction
Bug: b/225229697
Change-Id: I2d18b92bdef51362a9dbd9c0af56cb868e29869d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305462
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Linus Nilsson <lnilsson@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40121}
These tests verify the ability to override either the old or the
new function, and get the expected results.
Bug: webrtc:13931
Change-Id: Iebd0c929eda73dea75f32b96eb91a64e059a3cf8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294880
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40120}
StreamDataCounters is used both for send-side and receive side stats,
but last_packet_received_time is only used by receive statistician where
it duplicates another member
Bug: webrtc:13757
Change-Id: Iae6a65aba497e577ee3255e40623362e8c4c8a72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306183
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40119}
Finally remove bogus code after a year of no feedback on the matter.
Bug: webrtc:14138
Change-Id: I8083c9e1986e3779c9023a7d8935b717f63f0d86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306180
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40115}
This CL removes histograms which are no longer of use.
Bug: chromium:1255737
Change-Id: I7eb7e2cfbb03126257b51bfaa30d764b37dd9bd0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306200
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40114}
And fix bug that prevented it from passing.
Bug: webrtc:13931
Change-Id: I6cbc8e3aad704f6f7e33362efb7ec589ca6e6568
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306184
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40112}
This makes the handling somewhat more uniform, and is the same
for both video and audio channels.
Bug: webrtc:13931
Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40107}
Change default value of is_active to false,
this means that VideoRenderer or other VideoSinks
added with default rtc::VideoSinkWants() does not
block usage of RequestedResolution, e.g JNI_VideoTrack_AddSink.
This problem occurs when attaching a VideoRenderer directly to
the sending VideoTrack (which is a great solution!). But the
VideoRenderer is "passive" and should not block adaptations
from RequestedResolution.
Bug: webrtc:14451
Change-Id: I2ab02596245c7b82bf94fe86f8788f458c7ea286
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305024
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40105}
Reasons:
* Old code that has not been updated or tuned in several years.
* Nobody seems to intentionally use it.
* The application can do this itself by looking at GetStats.
Bug: None
Change-Id: Ib34bbebcf5885cf41ba05036506b500db7eb6b69
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306160
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40104}
Create a copy of flexfec_header_reader_writer for changing the implementation according to updated RFC. The fork is needed, since the updated RFC is incompatible with flexfec-03.
In the updated RFC, we receive the list and the number of protected ssrcs from the RTP header (from it's CSRCs , and CSRC count fields).
This Change is only a copy of the existing files. This will make it easier to understand the changes to the implementation in the next change sets.
Bug: webrtc:15002
Change-Id: I31bf5eca0d8f3cb23b4caabb477897eeb0ca6d96
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303240
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40103}
which is the maximum allowed in RFC 3550:
The last octet of the padding contains a count of how
many padding octets should be ignored, including itself
SRTP encryption does not need to be taken into account since none of
the cipher suites used by WebRTC require padding:
https://www.rfc-editor.org/rfc/rfc3711#section-3.1https://www.rfc-editor.org/rfc/rfc7714#section-7.2
BUG=webrtc:15182
Change-Id: Ife3d264af389509733699f2dd4d32ba63793e9de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305642
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40101}
Error resilience is no longer required for upper temporal layers.
Disabling error resilience on the upper layers leads to a ~2% PSNR BD-rate gain.
Reland of https://webrtc-review.googlesource.com/c/src/+/302001
Bug: webrtc:15106
Change-Id: I72ca9d504a7848dda934cbd52669027061742256
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305782
Reviewed-by: Jerome Jiang <jianj@google.com>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@google.com>
Cr-Commit-Position: refs/heads/main@{#40099}
