RtpPacket::UpdateDelayStatistics was previously optimized with several
sanity checks added. These sanity checks caused many of the unit tests
in peerconnection_integration_unittests to fail and the CL was therefore
reverted. This CL contains the sanity checks along with patches so that
the unit tests pass.
Bug: webrtc:9439
Change-Id: Ia5f5e8b125e5f3f4b79d433e2282901143530a25
Reviewed-on: https://webrtc-review.googlesource.com/99802
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24813}
This avoids crashing if an application resets the starting bitrate
before adding streams to the call.
Bug: webrtc:9586
Change-Id: I8d31aba1f4fee40c67c8930f5a32d17700ccadc3
Reviewed-on: https://webrtc-review.googlesource.com/101680
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24808}
The proper signature of the function should return an RTCError,
but due to all the classes in external projects implementing it with
this signature, we need a many steps process to update this.
- Add an implementation for the pure virtual method
- Update projects not to override it
- Update the function signature
- Update projects to override it with the right signature
- Remove the dummy implementation from the interface
Bug: webrtc:9777
Change-Id: Idf99216792b4ad13339e4e8be6f7b735bb6b64e7
Reviewed-on: https://webrtc-review.googlesource.com/101564
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24807}
Add helper class to process RtcEventLog events in order.
Use helper class to migrate rtc_event_log2rtp_dump.cc
to new parser API.
Bug: webrtc:8111
Change-Id: I7cbc220dad1f50be3a985ed44de27b38e5f20476
Reviewed-on: https://webrtc-review.googlesource.com/98601
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24806}
Use mock.patch instead of setattr, deemed hackish and less robust.
As an additional benefit, mock is explictly activated and precisely scoped.
Bug: chromium:855108
Change-Id: I3644bb6773a4b95e50aa5b671292e108af1fd2e9
Reviewed-on: https://webrtc-review.googlesource.com/101660
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24804}
It will soon lose the ability to do so.
Also, the ACM no longer creates comfort noise encoders for us, so
don't bother testing that.
Bug: webrtc:8396
Change-Id: I24a12e26bef142f9f8e7532b764f28572e0c6ace
Reviewed-on: https://webrtc-review.googlesource.com/101640
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24803}
Speeds up adaptation of the matched filter of the delay estimator by
allowing the estimated echo and the error signal (microphone minus
estimated echo) to be saturated. Only microphone saturation pauses
the filter adaptation.
Bug: webrtc:9773
Change-Id: I8b8400539fde3ee821f36a95818bece02ddd626b
Reviewed-on: https://webrtc-review.googlesource.com/101341
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24802}
When two-byte header extensions are enabled, kMaxId will change from 15
to 255. This CL is a refactor to remove the direct dependency between
memory allocation and kMaxId.
Bug: webrtc:7990
Change-Id: I38974a9c705eb6a0fdba9038a7d909861587d98d
Reviewed-on: https://webrtc-review.googlesource.com/101580
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24801}
A (actually several weeks) while ago, we noticed an error with the
WebRTC.Audio.Agc2.EstimatedNoiseLevel histogram. It always reported
the value 0. Here is why:
The histogram bins go from 0 to 100. But the value logged is dBFS. It is
always less than or equal to 0. This CL changes the bins.
Bug: webrtc:7494
Change-Id: I45fd122e98f9396f9871bc965a708987bd1815f6
Reviewed-on: https://webrtc-review.googlesource.com/101340
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24800}
This reverts commit 89b2963810b4cea0f95abdce011cb4e12fcdf1a1.
Reason for revert: Make experiment default off to not mess up data in re-launch.
Original change's description:
> Reland "Enable simulcast screenshare by default"
>
> This is a reland of d43c692ba7f53b5576a494c0343bc7a4bb36831b after fixes
> to failing chromium tests. No change to the original CL were done.
> Original CL reviewed on: https://webrtc-review.googlesource.com/87560
>
> TBR=stefan@webrtc.org
>
> Bug: chromium:690537
> Change-Id: I6b59ffc90d789aff21c7e52b118d3dfbe756c8a9
> Reviewed-on: https://webrtc-review.googlesource.com/89081
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24013}
TBR=ilnik@webrtc.org,sprang@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:690537, b:116052898
Change-Id: I429677de5547ce3a7badfb4414231ff9589e7414
Reviewed-on: https://webrtc-review.googlesource.com/101560
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24798}
Right after a volume decrease, the echo path estimate is overestimated and, as a side effect, the nearend signal is also overestimated. Due to that, the suppression gains are kept high avoiding the suppression of echoes. In this CL the neared power spectrum estimation is limited to a level given by the power spectrum or the microphone input signal. Additionally, the minimum gain that is computed inside the suppressor is also modified. Instead of using the nearend power spectrum that is now bounded, the power spectrum of the signal after the linear echo canceler is used.
Bug: webrtc:9762
Change-Id: Ia24cd2ce248f2c2ba124711b75acff3b8c5cfa9f
Reviewed-on: https://webrtc-review.googlesource.com/100720
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24796}
Deleted methods HttpData::setContent and
HttpData::setDocumentAndLength, as well as the
StreamInterface::GetAvailable method which becomes unused.
Bug: webrtc:6424
Change-Id: I6f360b68327d5964b2a18a9c4055255d774f6cbc
Reviewed-on: https://webrtc-review.googlesource.com/101180
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24793}
Replace calls to .str() which copies with .Release which moves in cases where that's safe.
This CL was generated by this command:
git grep -l 'StringBuilder' |
xargs perl -i -0 -pe "s/(rtc::StringBuilder (\S+);.*?return )\\g2.str\(\)/\$1\$2.Release\(\)/sg"
Bug: webrtc:8982
Change-Id: If4dadbeb039df010aaaa9e58da81c1971a84fe8f
Reviewed-on: https://webrtc-review.googlesource.com/100307
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24790}
The function checks the file extension to determine YUV or Y4M format.
Also adds a flag aligned_output_file to compare_videos.py, which allows
saving the aligned reference video to a file.
Bug: webrtc:9642
Change-Id: Ia59f5c123a1e41104756eb6b235b6581c4ffbd77
Reviewed-on: https://webrtc-review.googlesource.com/99503
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24787}
to satisfy a stricter check introduced in
503174a3e1
The file is supposed to contain actual gtest results, so having an
empty one is a workaround, but this just returns things to the way
they were.
TBR: phoglund@webrtc.org
No-Try: True
Bug: webrtc:9767, chromium:885194
Change-Id: I693cc2df9dfcafd7b728deb9efd445d8fe2c4edf
Reviewed-on: https://webrtc-review.googlesource.com/101301
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24783}
This is to help debug a depot_tools auth problem and can be reverted
once it is solved.
Bug: skia:8394
Change-Id: I3c713fce6c6ba6edbd2498d95938b48a28eff588
Reviewed-on: https://webrtc-review.googlesource.com/101160
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Commit-Queue: Eric Boren <borenet@google.com>
Cr-Commit-Position: refs/heads/master@{#24781}
It's ok for some WebRTC dependencies not to be Chromium dependencies.
Explicitly list them instead of relying of CIDP discrimination.
Bug: chromium:855108
Change-Id: I2dafce488b28409cbce7e0c3167d92f48859084f
Reviewed-on: https://webrtc-review.googlesource.com/101000
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24774}
Until now, only revision changes were automatically rolled.
This CL detect new and removed dependencies in third_party/android_deps.
Change-Id: I4f83b7308be577115cc3ed57edd9881496428173
Bug: chromium:855108
Reviewed-on: https://webrtc-review.googlesource.com/100021
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24773}
VP9 frame rate controller is supposed to be used in screen mode only
but it was partially enabled in normal video mode. This restricts use
of VP9 frame rate controller to screen mode.
Bug: chromium:884164
Change-Id: Ie2eaa31f3364a8abccbc4171007708cf7040fc38
Reviewed-on: https://webrtc-review.googlesource.com/100424
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24769}
This makes it clearer which code actually depends on the RelayPort
class.
Bug: None
Change-Id: I7b88de1824d5b5832d2f35a8820c5c59d05441c2
Reviewed-on: https://webrtc-review.googlesource.com/100801
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24768}
When simulcast screenshare is enabled, the max bitrate for the high
quality stream can be limited based on the resolution.
This CL fixes a bug where that limit could get so low that it is below
the min bitrate of the top layer, which in turn could cause the encoder
to fail initialization.
Bug: webrtc:9761
Change-Id: I093bd0ba68fe0165e8982d169daf02cdf912c924
Reviewed-on: https://webrtc-review.googlesource.com/100682
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24767}
This is a reland of 55daf1aef65218a97eff88999e5190a2f2f6b72e.
In order to avoid problems on case insensitive file systems this CL
moves rtc_export.h to rtc_base/system (avoiding problems with build/).
Diff: https://webrtc-review.googlesource.com/c/src/+/100804/1..2.
Original change's description:
> Add RTC_EXPORT macro to export WebRTC symbols.
>
> This CL introduces the utility macro RTC_EXPORT which will let WebRTC
> developers decide which symbols are supposed to be exported/imported
> and which ones are private.
>
> RTC_EXPORT will only export/import symbols in a component build, more
> info: https://cs.chromium.org/chromium/src/docs/component_build.md.
> During a component build, the macro COMPONENT_BUILD will be globally
> defined in a consistent fashion so it is safe to rely on it to
> understand how to expand RTC_EXPORT.
> In a non component build, RTC_EXPORT will expand to nothing.
>
> Bug: webrtc:9419
> Change-Id: Ic58162783be7f5883136ade27f324d6d34fdf932
> Reviewed-on: https://webrtc-review.googlesource.com/97960
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Yves Gerey <yvesg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24757}
Bug: webrtc:9419
Change-Id: Icfedea5fc3416ea1af2185de443fa879fb2dee8b
Reviewed-on: https://webrtc-review.googlesource.com/100804
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24766}
