This is comparable to this change done previously for for audio streams:
https://webrtc-review.googlesource.com/c/src/+/222042
This is a reland of commit 16a8b25d809e4d4982f9fc4b4e973acd506b8bca
with an additional fix in Patchset 2. Another problem turned out to be
in RTCPReceiver, which is fixed in:
https://webrtc-review.googlesource.com/c/src/+/262663
Bug: webrtc:11993
Change-Id: I63c7cf62a6dd50f88b491fea3ba866697552ef5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262665
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36907}
VP9 automaticResizeOn is disabled if more than one spatial layer is configured via scalability mode.
Bug: webrtc:13960
Change-Id: I7c6351bca6d2f32bcc7391894e8dcc9e74ca2050
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261315
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36897}
This reverts commit 16a8b25d809e4d4982f9fc4b4e973acd506b8bca.
Reason for revert: Checking if this is blocking the Chromium autoroller.
Original change's description:
> Update local_ssrc without needing to recreate video streams.
>
> This is comparable to this change done previously for for audio streams:
> https://webrtc-review.googlesource.com/c/src/+/222042
>
> Bug: webrtc:11993
> Change-Id: Ic953f816a8f7c56d1c3dc9a16d85bef3696a663d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261960
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36876}
Bug: webrtc:11993
Change-Id: I3a8d2f6a7e89b6784754d8e891a4e01479807c2d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262422
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#36892}
GetRtpExtensions() is still used in one corner case for audio receive
streams, so GetRtpExtensions has migrated to AudioReceiveStream.
Updated FlexfecReceiveStream config management (incl. pass by value) and
now store an RtpHeaderExtensionMap in FlexfecReceiveStreamImpl.
Call GetRtpExtensionMap() from call.cc instead of constructing one on
the fly for each rtp packet (for video packets at least).
Bug: webrtc:11993
Change-Id: Id90ec5d43ea368f58edd6f17cb39d8c54aec641f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261800
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36839}
This is to avoid accessing the array via the config struct.
Moving forward we might want to consider using the RtpHeaderExtensionMap
instead of a std::vector of RtpExtension.
Bug: webrtc:11993
Change-Id: I8469dbbd9bb95a69f87b5912bfc4bf8b8f603beb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261317
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36820}
`remote_ssrc` can be considered const while some other state represented
by rtp_config() can not and also is tied to a specific thread.
Separating access to these variables, makes moving things around easier.
Bug: webrtc:11993
Change-Id: I70aa000daab6174a401e01dca163213174e8f284
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261316
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36818}
Intended to let Vp8TemporalLayersFactory (an api/ target) reuse
this function, without depending on the codec implementation, and
without introducing a dependency cycle with the webrtc_vp8 build
target.
Bug: webrtc:11607
Change-Id: I671422e994e1005da8c7d768e8dd8ff795553e51
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261308
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36816}
This reverts commit 45361f78ed18c350b3edcaef19ae4c7cf167e95b.
Reason for revert: Perf alerts galore.
Original change's description:
> Calculate video stream max bitrate using expression.
>
> This replaces the ealier table-based caps.
> Apart from the VGA cap (now 1600kbps instead of 1700kbps), or if using
> "in between" resolutions, the caps are unchanged - but now cover high
> resolutions better.
>
> Bug: webrtc:14017
> Change-Id: I8649b528495d6c917e38ea8cb1a272df6c464c03
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260940
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36776}
Bug: webrtc:14017
Change-Id: I18ebc81c6054713c58d49bd227e37090686958c9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261309
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#36794}
This replaces the ealier table-based caps.
Apart from the VGA cap (now 1600kbps instead of 1700kbps), or if using
"in between" resolutions, the caps are unchanged - but now cover high
resolutions better.
Bug: webrtc:14017
Change-Id: I8649b528495d6c917e38ea8cb1a272df6c464c03
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260940
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36776}
Recent Clang versions have enhanced -Wunused-but-set-variable which now
warns about this.
Bug: chromium:1309955
Change-Id: Ie70df85f5a6d2cbabb4e10960bfd926ff7bf32cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257162
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Hans Wennborg <hans@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36381}
`enable_non_sender_rtt_` is updated a few lines below.
This seems to have been missed in the code review.
Bug: webrtc:12951, webrtc:13853
Change-Id: Idc06421362f6b2d831b5a828f296142aab9a46e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256860
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36380}
This patch takes a stab at modules/video_coding,
but reaches only about half.
Bug: webrtc:10335
Change-Id: I0d47d0468b818145470c51ae4e8e75ff58d499ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256112
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36335}
Motivation: never used.
Follow-up CL of https://webrtc-review.googlesource.com/c/src/+/250680.
Tested on a custom chromium build by making a video call and with
keyboard activity. The expected logs from the APM transient suppressor
sub-module were shown.
Bug: webrtc:11226
Change-Id: I4186994412dd8ba2e71ed8f9dcc9cf8f8e40fbd3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250667
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36300}
convert almost all of video/ (and the collateral)
Bug: webrtc:10335
Change-Id: Ic94e05937f54d11ee8a635b6b66fd146962d9f11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254601
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36192}
this has been enable by default since M96
BUG=webrtc:11640
Change-Id: I5d310d3929882007211eae12bc3ac1366107ca87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253400
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#36123}
Move complexity parameter to the main VideoCodec class to enable
additional video codecs to use the parameter without creating a new
codec-specific structure.
Bug: webrtc:13694
Change-Id: Icb7cf640b178875d799f39ade8b5084e3222bb1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251921
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Michael Horowitz <mhoro@google.com>
Cr-Commit-Position: refs/heads/main@{#36040}
PS#1 is a reland of "Added support for H264 YUV444 (I444) decoding." https://webrtc-review.googlesource.com/c/src/+/235340
Bug: chromium:1251096
Change-Id: Icd997c7f7732229954d5494343b4e7a70deb09d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251303
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35964}
This collision can occur when we have
asymetrical send and receive codecs. This is the case in the current
code base with the VP9 codec familly but is not visible untill more
codecs are added.
Added Nutanix Inc. to AUTHORS.
Bug: chromium:1291956
Change-Id: I09d3f76161d984d2a3edf721639753bffd4947b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250034
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35944}
This CL removes even more top-level const from parameters in function
declarations. This change is safe because top-level const in function
declarations (not function definitions) are ignored by the compiler
and so change is just a no-op cleanup.
Bug: webrtc:13610
Change-Id: Icf6868c27b1fdb9d9915b3a7020eb34bdcf07a09
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249989
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35866}
This reverts commit 3babb8af238a531cbff27951604b09bb78b762cd.
Reason for revert:
- Causes regressions to transceivers, see https://crbug.com/1291956 for more information, including tests to reproduce the issue.
This CL is not a pure revert. While it reverts everything else, it does
keep the new enum value (kProfilePredictiveHigh444). This is as to not
break Chromium which already depend on it. It is not listed in the
kProfilePatterns though so the enum value should never be applicable.
Original change's description:
> Added support for H264 YUV444 (I444) decoding.
>
> Added Nutanix Inc. to the AUTHORS file.
>
> PS#1 is a reland of "Added support for H264 YUV444 (I444) decoding." https://webrtc-review.googlesource.com/c/src/+/234540
>
> Bug: chromium:1251096
> Change-Id: I99a1b1e4d8b60192ff96f92334a430240875c66c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235340
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35684}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:1251096, chromium:1291956
Change-Id: Ib4d8ea4898f9832914d88e7076e6b39da0c804ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249791
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35835}
The interface is implemented by the ChannelManager and contains methods
to create and destroy media channel objects as used by a transceiver.
This will subsequently allow us to delete the channel objects from
the transceiver class where ownership really lies rather than from
the outside - which is currently required by some tests that keep
channel objects on the stack. We'll furthermore be able to do the
destruction asynchronously without additional Invoke()s as we do now
which will remove an Invoke when making sdp changes.
With introducing the interface, the following simplifications were made:
* ChannelManager constructed on the signaling thread.
Before, there was an Invoke in the context class, which existed
for the purposes of calling MediaEngine::Init() (which in turn is
only needed for the VoiceEngine). This Invoke has now been moved
into the CM (more tbd).
* The CM now has a pointer to the signaling thread (since that's the
construction thread). That allows us to remove the signaling thread
parameter from the CreateFooChannel methods.
* The ssrc_generator (UniqueRandomIdGenerator) instance for SSRCs moved
from SdpOfferAnswerHandler to the CM, as it's always used in
combination with the CM. This simplifies the CreateFooChannel methods
as well as a couple of other classes that have a CM dependency.
* Removed DestroyFooChannel related code from SdpOfferAnswerHandler since
the channel type detail can be taken care of by the CM.
Bug: webrtc:11992, webrtc:13540
Change-Id: I04938a803734de8489ba31e6212d9eaecc244126
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247904
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35766}
Currently `CreateLibaomAv1Encoder` will either return an actual libaom AV1 encoder or a nullptr depening on whether the build flag `enable_libaom` was configured to true or not. This CL updates the `libaom_av1_encoder` build target to no longer depend on `enable_libaom` so that `CreateLibaomAv1Encoder` will always return an encoder instance.
Added `CreateLibaomAv1EncoderIfSupported` as a replacement to the old `CreateLibaomAv1Encoder`.
Bug: webrtc:13573
Change-Id: Ibdcd52c609acd79feefa2b86f19d1b4ca3e91d0a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242360
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35763}
Currently if encoder initialization fails WebRTC doesn't send any video.
This CL adds functionality that changes encoder type in such case and
restores the video. If encoder selector is available we switch to
encoder it recommends. Otherwise, VP8 is used as the default fallback
encoder.
Bug: webrtc:13572
Change-Id: Ifcdf707a575711f5ff81f9451caf30140c9171dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246960
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35761}
WebRTC can switch encoder on-fly when encoder fails or by request from
encoder selector. Putting the current send codec to the front of the
codecs list provides a simple way for apps to know what is actually
used without retrieving stats.
Bug: webrtc:13572
Change-Id: Iaaa5f7ad8667f59016dc92bff9e9a57a7425ef44
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246500
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35723}
This can cause issues on Android S if this initialization happens when
the app doesn't have permission to access the microphone.
Bug: b/197461765
Change-Id: Iebccff9d15f5bb12a7b2c78e1c373e379b37a127
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246104
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35689}
Added Nutanix Inc. to the AUTHORS file.
PS#1 is a reland of "Added support for H264 YUV444 (I444) decoding." https://webrtc-review.googlesource.com/c/src/+/234540
Bug: chromium:1251096
Change-Id: I99a1b1e4d8b60192ff96f92334a430240875c66c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235340
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35684}
Because of this (seemingly simple) change, I had to change the return
type of transport_name from `const std::string&` to `absl::string_view`
to handle the case when there's no transport assigned.
That in turn caused an avalanche of required updates.
Bug: webrtc:12230, webrtc:11993
Change-Id: I16ec6c6a5fc2f5f7c7de572355a3c6ca924bb9d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244084
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35617}
This makes things slightly simpler for the time being as surrounding
code is being refactored. This also removes a PostTask which has the
effect of shrinking the window between the Pending/Complete
notifications slightly since there's no additional async task
for the 'complete' step.
Bug: webrtc:11993
Change-Id: Ia86779b21c6f87301f37d763f89ace722e06e563
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244081
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35609}
