This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251, webrtc:163
Change-Id: I6f03c46e772ccf4d15951a4b9d4e12015d539e58
Reviewed-on: https://webrtc-review.googlesource.com/90408
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24113}
All clients who do not own their own APM turn it off by default
(in WebrtcVoiceEngine). AECM with comfort noise is a little-exercised
code path. Configurability of this setting is going away, so we're
better off disabling it by default.
Bug: webrtc:9535
Change-Id: Iba839aa18e79ae29ff20bdf6e30de77870ba4143
Reviewed-on: https://webrtc-review.googlesource.com/89583
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24078}
Move base64.h to the proper location and put redirect header into the
old place to be able to switch downstream users on new location.
Bug: webrtc:8366
Change-Id: I5191fe631d32178d2efd1315ca9abd4250102291
Reviewed-on: https://webrtc-review.googlesource.com/88223
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24069}
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251, webrtc:163
Change-Id: I5475e574353c772910181495fdb3400b5f0e7399
Reviewed-on: https://webrtc-review.googlesource.com/87240
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24040}
The current default behavior is unchanged and points to MAINTAIN_FRAMERATE,
meaning there is no way to currently use BALANCED as we can't detect
when the value as been set or not.
Updating this is an API change that should be done in another CL and
properly communicated first.
Bug: webrtc:7607
Change-Id: Ic3877ad8dd7bc418296f21a04bc37f59ec55934a
Reviewed-on: https://webrtc-review.googlesource.com/88766
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24024}
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251, webrtc:163
Change-Id: I6d3b45de9dca3a5a04f0cdd5583919d35a585a7e
Reviewed-on: https://webrtc-review.googlesource.com/89043
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24018}
Currently the codec specific max bitrate that is set in the SDP
gets overridden by the value set with the "b=AS" attribute
(WebRtcVideoChannel::SetSendParameters). But at the
WebRtcVideoSendStream level it does the opposite - the codec
specific max bitrate value overrides the values that could be
set by RtpParameters or the "b=AS" value
(in WebRtcVideoSendStream::CreateVideoEncoderConfig). This change
updates the logic to be consistent with what happens at the
WebRtcVideoChannel level, and allows the RtpParameter max bitrate
to override the codec specific max bitrate.
Bug: webrtc:8655
Change-Id: I3f0347cb7cffcfc577484231b061ab0712453e69
Reviewed-on: https://webrtc-review.googlesource.com/88520
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23989}
Reverses check for adding RTX to a codec. With this change RTX will
be added to external codecs.
Bug: webrtc:9516
Change-Id: Ie60b0b629dd9b05cbf20b2799bbf9bdccd8a6bcf
Reviewed-on: https://webrtc-review.googlesource.com/88441
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23962}
This is an reland of 6f5b0f920af08d66e6b77ee4f91ade5797145368
Relanded after speculative revert without any changes.
TBR=ilnik@webrtc.org
Original change's description:
> Remove rtc::Optional alias and api:optional target
>
> Update left-overs where old target still was used.
>
> Bug: webrtc:9078
> Change-Id: I2162c928091fc4ff1dea33a3f03adbe47207d206
> Reviewed-on: https://webrtc-review.googlesource.com/84740
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23913}
Bug: webrtc:9078
Change-Id: Ia33c6438253c6ec49f45d938e8a3607b51c418be
Reviewed-on: https://webrtc-review.googlesource.com/88160
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23941}
Prepare for building without built-in software codecs. When passing
null, inject the new type of factories but wrap them in the built-in
software codecs outside the videoengine.
Bug: webrtc:7925
Change-Id: I7408e6e46e6b9efdf346852954bf51a97e023b5c
Reviewed-on: https://webrtc-review.googlesource.com/83729
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23897}
Also adjust to base-layer fraction for the shortened 3-tl pattern to be
60%, just like the 2-tl setting.
This CL removes direct use of the allocation matrix and moves it behind
a static getter.
Bug: webrtc:9477
Change-Id: Ifd7d1edffa0555024fd252834357b926997d13b5
Reviewed-on: https://webrtc-review.googlesource.com/86681
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23834}
This CL adds VP9 profile information in SDP. It adds the necessary fields and
enums to codec containers.
Additional profiles will be followed.
Bug: webrtc:9376
Change-Id: I78574714f06f8087262a71dd64c01f31a229dd54
Reviewed-on: https://webrtc-review.googlesource.com/81960
Reviewed-by: Taylor (left Google) <deadbeef@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23810}
This will allow experimenting with audio bitrate allocation in video calls without increasing transport overhead.
Bug: webrtc:8243
Change-Id: If961780921d53bdce95b68c26641df6875509c1f
Reviewed-on: https://webrtc-review.googlesource.com/84501
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23755}
* Move SimulcastEncoderAdapter out under modules/video_coding
* Move SimulcastRateAllocator back out to modules/video_coding/utility
* Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility
* Move any VP8 specific code - such as temporal layer bitrate budgeting -
under codec type dependent conditionals.
* Plumb the simulcast index for H264 in the codec specific and RTP format data structures.
TBR=sprang@webrtc.org,stefan@webrtc.org,titovartem@webrtc.org
Bug: webrtc:5840
Change-Id: I2d3b130622dd7ceec5528f3ab6c46f109e6bafb8
Reviewed-on: https://webrtc-review.googlesource.com/84743
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23715}
This reverts commit 07efe436c9002e139845f62486e3ee4e29f0d85b.
Reason for revert: Breaks downstream project.
cricket::GetSimulcastConfig method signature has been updated.
I think you can get away with a default value for temporal_layers_supported (and then you can remove it after a few days when projects will be updated).
Original change's description:
> Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8.
>
> * Move SimulcastEncoderAdapter out under modules/video_coding
> * Move SimulcastRateAllocator back out to modules/video_coding/utility
> * Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility
> * Move any VP8 specific code - such as temporal layer bitrate budgeting -
> under codec type dependent conditionals.
> * Plumb the simulcast index for H264 in the codec specific and RTP format data structures.
>
> Bug: webrtc:5840
> Change-Id: Ieced8a00e38f273c1a6cfd0f5431a87d07b8f44e
> Reviewed-on: https://webrtc-review.googlesource.com/64100
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23705}
TBR=sprang@webrtc.org,stefan@webrtc.org,mflodman@webrtc.org,hta@webrtc.org,sergio.garcia.murillo@gmail.com,titovartem@webrtc.org,agouaillard@gmail.com
Change-Id: Ic9d3b1eeaf195bb5ec2063954421f5e77866d663
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:5840
Reviewed-on: https://webrtc-review.googlesource.com/84760
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23710}
* Move SimulcastEncoderAdapter out under modules/video_coding
* Move SimulcastRateAllocator back out to modules/video_coding/utility
* Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility
* Move any VP8 specific code - such as temporal layer bitrate budgeting -
under codec type dependent conditionals.
* Plumb the simulcast index for H264 in the codec specific and RTP format data structures.
Bug: webrtc:5840
Change-Id: Ieced8a00e38f273c1a6cfd0f5431a87d07b8f44e
Reviewed-on: https://webrtc-review.googlesource.com/64100
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23705}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
Extract rtc_base/base64.h and rtc_base/base64.cc into separate target
to prepare to move them into third_party
Bug: webrtc:8366
Change-Id: I477e6da2b9d09307439b3272261f31042f479d74
Reviewed-on: https://webrtc-review.googlesource.com/83980
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23645}
This was called by FakeWebRtcVideoCaptureModule's destructor.
However, since the factory keeps a reference counted pointer to
each FakeWebRtcVideoCaptureModule it has created, no module is
destroyed until the factory is destroyed. And at that point,
coordination is not needed and actually broken, since OnDestroyed
results in Release being called on an object halfway through the
destruction sequence.
Bug: webrtc:9405
Change-Id: I0cf3acb49b58e2b6b83344d840835f594418f7c7
Reviewed-on: https://webrtc-review.googlesource.com/83721
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23644}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameters 'media ortc p2p':
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I19167714af7cc1436d34cfcba6c8b3718d8e677b
Reviewed-on: https://webrtc-review.googlesource.com/83731
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23638}
This experiment is now wired up inside of chrome using field trial and
this passthrough is now obsolete.
Bug: chromium:794608
Change-Id: I1407e391d39c7e8696add9f656f059e7d8a27a08
Reviewed-on: https://webrtc-review.googlesource.com/82780
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23625}
