This reverts commit a1e4fbb25371867349a0c2ed6ba62224735a2ec7.
Reason for revert: Breaks downstream.
Original change's description:
> Improving robustness of feedback matching code in event log parser.
>
> Removes the dependency on TransportFeedbackAdapter thereby removing
> some of the complexity that came with it, in particular, we don't fill
> in missing packets. This makes the code easier to debug and avoids some
> confusing logging that's not relevant for the parser.
>
> Bug: webrtc:9883
> Change-Id: I6df8425e8ab410514727c51a5e8d4981d6561f03
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133347
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27739}
TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Change-Id: Icdf3231f5a32b6f63a903c7dffc8ca505680a72a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134105
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27742}
The latter is also a member of the former. This cleanup is also
a preparation for dropping WebRtcRTPHeader::frameType (or deleting
WebRtcRTPHeader right away), now that it's a video-specific member.
Tbr: kwiberg@webrtc.org # Comment change in modules/include/
Bug: None
Change-Id: I5c1f3f981f0d750713fc9b9b145278150fe32b5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133024
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27740}
Removes the dependency on TransportFeedbackAdapter thereby removing
some of the complexity that came with it, in particular, we don't fill
in missing packets. This makes the code easier to debug and avoids some
confusing logging that's not relevant for the parser.
Bug: webrtc:9883
Change-Id: I6df8425e8ab410514727c51a5e8d4981d6561f03
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133347
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27739}
LibvpxVp8Encoder::Encode() creates a local instance of
rtc::scoped_refptr<I420BufferInterface>, then sets members to
point into the internal state of that I420BufferInterface. These
pointers remain in place after the buffer is destroyed.
This CL fixes the issue by deleting the references when the
function exits.
Bug: webrtc:10570
Change-Id: I9623e2ff3dd43e8fd1d1cc7696a3f28227d4544b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133882
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27738}
This reverts commit 00d0a0a1a9520fb4323d7e3a1c02133b7b942978.
Reason for revert: Breaks downstream tests
Original change's description:
> Copy video frames metadata between encoded and plain frames in one place
>
> Currently some video frames metadata like rotation or ntp timestamps are
> copied in every encoder and decoder separately. This CL makes copying to
> happen at a single place for send or receive side. This will make it
> easier to add new metadata in the future.
>
> Also, added some missing tests.
>
> Bug: webrtc:10460
> Change-Id: Ia49072c3041e75433f125a61050d2982b2bec1da
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27719}
TBR=ilnik@webrtc.org,sprang@webrtc.org,kron@webrtc.org
Change-Id: I8960a6cc15e552925129ba0037f197ff3fd93c25
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10460
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134100
Reviewed-by: Artem Titarenko <artit@webrtc.org>
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27737}
In simulcast screenshare the lower stream can be disabled for ~2 seconds
due to bandwidth limitations. During that time with 30 input fps more
than 50 frames can be pending.
This CL remove unnecessary warnings.
Bug: webrtc:4172
Change-Id: I979c946a03ff3f67f500843c66382e437ecd559b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134041
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27735}
The function iterated over two containers, destroyed their elements
and popped those elements one at a time. It's more efficient to
destroy all of the elements, then clear() the container.
Bug: None
Change-Id: I17aa88694ee41df64c5793b08b96899b7ff04071
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133901
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27730}
Instead of crashing when encountering an event log that cannot be parsed
it is better to print an error message, skip the file and continue.
Bug: webrtc:10337
Change-Id: I5dbca18e456c14e5a92af068f82e88cb17e8de9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133185
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27727}
Currently some video frames metadata like rotation or ntp timestamps are
copied in every encoder and decoder separately. This CL makes copying to
happen at a single place for send or receive side. This will make it
easier to add new metadata in the future.
Also, added some missing tests.
Bug: webrtc:10460
Change-Id: Ia49072c3041e75433f125a61050d2982b2bec1da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27719}
The one with enum.IntFlag is not feasible. An attempt is done here:
https://webrtc-review.googlesource.com/c/src/+/133884
It requires re-writing QualityAssessment to Python3 which is too much
work for little benefit. (I tried, but couldn't get the unit-tests to
pass for both 2 and 3.)
The second one is not a real todo.
TBR=alessiob@webrtc.org
NOPRESUBMIT=True
Bug=None
NOTRY=True
Change-Id: Ia25817533cd504c30490f86e4058f0b2d59dd39c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133908
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27715}
With this change, both the normal RTP and the transport-wide sequence
numbers are propagated with with AddPacket() call via a new
RtpPacketSendInfo struct, replacing the previous set of parameters.
The intent with this is that SendTimeHistory can hold a mapping from
transport-wide to rtp sequence numbers, and then via callbacks let the
RTP modules know when packets have been received by the remote end.
Bug: webrtc:8975
Change-Id: I6a24fc6282cbb041393752d39593c2867b242192
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133021
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27708}
According to crash reports, crash happens at the line with nothing but
|next_frame->second.frame->is_last_spatial_layer|.
Probably, |frames_| contains entries with empty frame unique_ptr.
This CL adds checks to not dereference those empty pointers.
Bug: chromium:955040
Change-Id: I3060f9e1af8bfc3c8a079c14107b5b4a82f5d015
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133626
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27706}
They are called only from VideoReceiveStream, which can access
VCMTiming directly.
Bug: webrtc:7408
Change-Id: Ibf5799b1441c00b41143342ca1d99024cb68ba17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133569
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27700}
The acoustic echo canceler AEC2 is being deprecated. The routing for reporting these metrics as UMA stats has outlived the metrics'usefulness.
Bug: webrtc:10563
Change-Id: Ib96693dfc43e25a0cfecad7d5d2043116ca7e6b5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133573
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27699}
And fill-in icc profile from the various window and screen capturers.
Done on WindowCapturerMac, ScreenCapturerMac, WindowCapturerX11
and ScreenCapturerX11. Follow-up CLs will do it on ScreenCapturerWinDirectx
and ScreenCapturerPipeWire.
Useful to build the gfx::ColorSpace in chromium, especially
from src/content/browser/media/capture/desktop_capture_device.cc.
We do not build the color space directly here to avoid duplicating
ui/gfx/icc_profile.h,cc code from chromium, which one implements
icc profile caching.
Bug: chromium:945468
Change-Id: Id6e3920233771e035f7578847406bf1f519dcd49
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133580
Commit-Queue: Julien Isorce <julien.isorce@chromium.org>
Reviewed-by: Sergey Ulanov <sergeyu@chromium.org>
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#27697}
By wraping the cg_data instead of copying it. We had the infrastructure
for it since the work around iosurface, we were just not using it.
Also having a centralized DesktopFrameCGImage::CreateFromCGImage helper
will be useful to parse the ICC Profile at only one place.
Bug: chromium:945468
Change-Id: I69f179064fd9045d992a7baea35820c38e24dacc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133640
Commit-Queue: Julien Isorce <julien.isorce@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#27696}
- add ParseOutgoingBitstreamAndRewriteSps to SpsVuiRewriter
which takes encoded H.264 bitstream and NAL unit boundaries,
rewrites SPS if needed and updates the NAL unit boundaries
accordingly
- move SPS rewriting stats updates to SpsVuiRewriter
Bug: webrtc:10559
Change-Id: I7ca21756628ee6d6abbcbd501bdb4f3df024168b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133174
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27665}
The idea is that when ALR is detected, the encoder can not produce the bitrate
needed for the delay based estimator to detect overuse and thus the delay based
estimator should not be allowed to increase further.
Likewise, if ALR is not detected, the delay based estimator is allowed to
increase the BWE to ensure that there is no region where the BWE can get stuck.
BUG=webrtc:10542
Change-Id: Ic94b708461c9077fd09132ee4ecb6279ffcd5f99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133190
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27661}
Rename "UpdateLayerConfig" to the more appropriate "NextFrameConfig".
Also update some comments in vp8_frame_buffer_controller.h.
Bug: None
Change-Id: Iba8227f84e33e5ebd28d2eeb10fe03e776036603
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133202
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27660}
A typo in a previous CL made OnLossNotification() accept its
single argument as a const-value, rather than a const-reference.
Bug: webrtc:10501
Change-Id: I5e6f9c79f15205b75ec90a53d3fccf3dd9927e33
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133343
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27659}
We knew that we should not update buffer level during DTX period. We already fulfill this upon no packet receipt. But we missed doing it for DTX-signaling packets. This CL is to fix that.
Bug: b/129521878
Change-Id: I72ca18e3b21e956123fe6e3119ef0d7c981c9eec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133183
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27643}
Instead use WebRtcKeyValueConfig and FieldTrialBasedConfig
BUG=webrtc:10335
Change-Id: Ie148cb466f86d8fa1ded5c7f125fbcccf6e7dbe3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132714
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27642}
As non-linear mode uses a suppressed version of y (not e) as output, this change
uses Y2, rather than E2, as nearend spectrum when computing the suppression gains.
E2 is still used in linear mode.
This change also affects how the minimum suppression gains are calculated. The
minimum gain is now min_echo_power / weighted_residual_echo.
Bug: webrtc:10550
Change-Id: I2904c5a09dd64b06bf25eb5a37c18dab50297794
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133023
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27629}
Prior to this CL, this was indicated by passing |size_bytes| = 0
to the method.
Bug: webrtc:10501
Change-Id: Icff3bb83344834dc62d62bde5ec5d05096a08e11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132712
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27620}
Instead use WebRtcKeyValueConfig and FieldTrialBasedConfig.
The purpose is to allow a user of GoogCC to use different settings on different instances.
BUG=webrtc:10335
Change-Id: I2f837688c9fdd341eecb44484cc784b1c80da1a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132791
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27617}
This allows picking up the output in Android tests, where stdout/stderr
is lost but RTC_LOGs are picked up by the org.webrtc.Logging utility.
Tested: Downstream Android tests.
Bug: webrtc:10349
Change-Id: I1379f4303640dbc9621c64d9c88cf61bc8447ab6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132704
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27616}
This CL adds an experiment where aggressiveness of the rate controller
is tuned based on if the application is network constrained or not.
Bug: webrtc:10155
Change-Id: I6c8cd116f57321c5b36cf5a69840913936091aaa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132786
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27615}
