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}
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}
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}
The previous attempt caused issues:
https://webrtc-review.googlesource.com/c/src/+/86900
Let's try it with a separate file instead.
Bug: None
Change-Id: I57dc4dceca1ea4b73003d4202e9b75ee469e5adc
Reviewed-on: https://webrtc-review.googlesource.com/86940
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23820}
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}
Target bitrate is set to 0.75 of the max bitrate.
Bug: webrtc:9341, webrtc:8655
Change-Id: I9a8c8bb95bb1532d45f05578832418464452340e
Reviewed-on: https://webrtc-review.googlesource.com/79821
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23676}
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}
Need to depend on them from Chromium.
Bug: webrtc:7925
Change-Id: Iea1bb3b937c602920bfd87f885c87c790ac7bc17
Reviewed-on: https://webrtc-review.googlesource.com/82061
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23580}
And update most internal calls to use it.
Bug: webrtc:5740, webrtc:9372
Change-Id: Ib57d4ebfa7b0729af6d22981a792f0fdadf8a13f
Reviewed-on: https://webrtc-review.googlesource.com/81743
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23567}
The proper closing procedure is:
1. Alice resets outgoing stream.
2. Bob receives incoming stream reset, resets his outgoing stream.
3. Alice receives incoming stream reset; channel closed!
4. Bob receives acknowledgement of reset; channel closed!
https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#section-6.7
However, up until now we've been sending both an incoming and outgoing reset
from the side initiating the closing procedure, and doing nothing on the remote
side.
This means that if you call "Close" and the remote endpoint is using an old
version of WebRTC, the channel's state will be stuck at "closing" since the
remote endpoint won't send a reset. Which is already what happens when Firefox
is talking to Chrome.
This CL also fixes an issue where the DataChannel's state prematurely went to
"closed" before the closing procedure was complete. Which could result in a
new DataChannel attempting to re-use the ID and failing.
TBR=magjed@webrtc.org
Bug: chromium:449934, webrtc:4453
Change-Id: Ic1ba813e46538c6c65868961aae6a9780d68a5e2
Reviewed-on: https://webrtc-review.googlesource.com/79061
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23478}
This will allow exposing the interface to downstream users that
want to test VP8 simulcast. No functional changes to the tests
themselves are expected.
Bug: webrtc:9281
Change-Id: I4128b8f35a4412c5b330cf55c8dc0e173d4570da
Reviewed-on: https://webrtc-review.googlesource.com/77361
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23469}
Experiment flag added to PeerConnectionInterface::RtcConfiguration and
propagated down to VideoStreamEncoder.
Artificial Sdp parameter is added to the sdp format if the flag is set.
Additionally, sdp format is propagated in vp8 simulcast adapters.
Bug: chromium:794608
Change-Id: I2dec54d19ae7bfbd5f2777ec682da5a84194da94
Reviewed-on: https://webrtc-review.googlesource.com/78500
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23412}
