Also initial implementation wrapping the libaom AV1 encoder.
Note that for now this is intended for prototype purposes.
Bug: none
Change-Id: Iac42ca4aecb6a204601c9f00bfb300e3eda3c4f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306181
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42108}
This removes many references to the unsupported ProxyInfo struct
but leaves temporary implementations for methods while downstream
code gets updated.
Bug: none
Change-Id: Iab4410b362a8296b2e00cf71080010e515f9f4ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344660
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42096}
This reverts commit 501c4f37bfee47b26999ee291c5355ad64554df7.
Patch set 1 contains pure reland.
The reason why we want to do this is because audio can allocate a needed bitrate before video when starting a call, which may lead to a race between the first probe result and updating the allocated bitrate.
That is the, initial probe will try to probe up to the max configured bitrate.
Bug: webrtc:14928
Change-Id: I6a8660da20ac54237f04a29461e03b31bd988bb0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347643
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@google.com>
Owners-Override: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42086}
Since iwyu is now deprecated, we need to enable use of include-cleaner.
This approach gives some error messages when running, but does the job.
Bug: webrtc:15874
Change-Id: I431deef0f2e5ce99eb256a4d82aa32769ae58b41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347642
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42085}
This reverts commit 33cc83595ae7dd144c57c614fb62d286d9d7bf68.
Reason for revert: Perf bots showed that this cl cause a change in metrics. It looks like it is for the better, but we want this to be behind a field trial.
Original change's description:
> Ignore allocated bitrate during initial exponential BWE.
>
> The reason why we want to do this is because audio can allocate a needed bitrate before video when starting a call, which may lead to a race between the first probe result and updating the allocated bitrate.
> That is the, initial probe will try to probe up to the max configured bitrate.
>
> ProbeController::SetFirstProbeToMaxBitrate will allow the first probe to
> continue up to the max configured bitrate, regardless of of the max
> allocated bitrate.
>
> Bug: webrtc:14928
> Change-Id: I6e0ae90e21a78466527f3464951e6033dc846470
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346760
> Reviewed-by: Diep Bui <diepbp@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42049}
Bug: webrtc:14928
Change-Id: I56ba58560b6857b6069552c02df822691f7af64d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347622
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Owners-Override: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42081}
This expose a new GetSupportedH265Level API for WebRTC external
factories to calculate H.265 levels to be use for SDP negotation.
Bug: webrtc:13485
Change-Id: Ib420da2b9b1b7af00129294be5b3efec172e8faf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/345544
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42079}
The reason why we want to do this is because audio can allocate a needed bitrate before video when starting a call, which may lead to a race between the first probe result and updating the allocated bitrate.
That is the, initial probe will try to probe up to the max configured bitrate.
ProbeController::SetFirstProbeToMaxBitrate will allow the first probe to
continue up to the max configured bitrate, regardless of of the max
allocated bitrate.
Bug: webrtc:14928
Change-Id: I6e0ae90e21a78466527f3464951e6033dc846470
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346760
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42049}
It is a public interface and must be visible to allow tests to include the header file.
Bug: none
Change-Id: I4e6322c622f62c018b274b751e2c395eed7816e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346520
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42027}
The predefined SdpVideoFormats were not used everywhere,
which caused a discrepancy between send/receive capabilities
for AV1. This CL solves the immediate problems by making sure
send/receive capabilities for AV1 are reported the same way.
Fixed: chromium:331565934
Change-Id: I073091b7b5f987c7f434c17276fd84047ec723c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344681
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41991}
RtpPacketToSend::transport_sequence_number
packed_id is set to be 64 bit to align with rtc::PacketOptions.
packet_id is only set to RtpPacketToSend::transport_sequence_number if
TransportSequenceNumber header extension is not used in order to not
change current behaviour.
Bug: webrtc:15368
Change-Id: Ia532714226421422bdb292f8dd34b175560e9dc6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344160
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41950}
Start migrating away from `hasAudioLevel`, `voiceActivity`, `audioLevel` fields in RTPHeaderExtension and switch usages to a more modern absl::optional<AudioLevel> accessor instead.
The old fields are preserved for compatibility with downstream projects, but will be removed in the future.
Bug: webrtc:15788
Change-Id: I76599124fd68dd4d449f850df3b9814d6a002f5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336303
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41947}
This code was extracted to make the next following CL easier to review.
This CL simply exposes the getters, setters and callbacks to set the
buffered amount low threshold on a specific SCTP stream. It will be
used in a follow-up CL, but is just boilerplate.
Bug: chromium:40072842
Change-Id: Iccd72208b369ddc252cc5886f6446b9c2ceeb0b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343360
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41943}
Before this change, calling buffered_amount only included what was
buffered on top of what was already buffered in the SCTP socket. With
the defaults, the SCTP socket can buffer up to 2MB of data (that is not
put on the wire) before the additional external bufferering in
SctpDataChannel will be used. The buffering that I am working on
removing completely.
Until it's removed completely, to avoid the issue reported in
crbug.com/41221056, include the bytes buffered in the SCTP socket to
what is returned when calling RTCDataChannel::buffered_amount.
This means that when this value is zero, it can be safe to know that all
bytes have been sent, but not necessarily acknowledged. And calling
close will not discard any messages.
This is a stopgap solution, but as functional as the proper solution
that removes all additional buffering. Follow-up CLs will merely improve
this solution.
Bug: chromium:41221056
Change-Id: I06edd52188d3bf13a17827381a15a4730722685a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342520
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41898}
Also remove all dependencies on rtc_media_base except for a few
that are suspected of being linker directives.
Bug: webrtc:14775
Change-Id: Ic0daf88b5422047d3ed7079ee6af9e689853310c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341461
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41886}
These lines cause an error when building a project with libwebrtc as a dependency in Microsoft Visual Studio.
Bug: webrtc:15864
Change-Id: I1abfe257d0ea1c16c4c5b718594e8085036f7763
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342320
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41881}
This reverts commit ed8390d21a7b15091d01bc8e843193d0a6efd23a.
Reason for revert: Fix has landed in chrome, ready to reland.
Original change's description:
> Revert "Deprecate old constructors and set_type() in Candidate and Port"
>
> This reverts commit aaa6851d53741179a591d79fc82c4dd6651a7ba5.
>
> Reason for revert: breaks chromium webrtc import
>
> Original change's description:
> > Deprecate old constructors and set_type() in Candidate and Port
> >
> > * Deprecates constructors that use string based `type`
> > * Deprecates string based type functions in favor of enum based.
> > * Restrict possible values of Candidate::type. Ensure a valid value
> > is assigned at construction.
> > * Make Port constructors protected to limit their use to subclasses.
> > - The reason for this is to make sure that use of SharedSocket()
> > is controlled (it adds a bit of complexity).
> > * Simplify construction of Port (remove Construct() etc)
> >
> > Bug: webrtc:15846
> > Change-Id: If24ed674e175642efa49da37fd2bc847dd14f613
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339860
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#41865}
>
> Bug: webrtc:15846
> Change-Id: Ic8b7cba97f8fb207ef51a88900e704658ade28b7
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342140
> Auto-Submit: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Owners-Override: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Cr-Commit-Position: refs/heads/main@{#41867}
Bug: webrtc:15846
Change-Id: I3d52643bbb537d1c072643528828d26eb18fea94
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342200
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41875}
The DcSctpTransport will soon use field trials to conditionally enable
some options.
And overall, there is a migration project to start using the Environment
and this CL is in that direction, also setting the boundary; The dcSCTP
library should not depend on it. But the transport is allowed to.
Bug: webrtc:14997
Change-Id: I1f3c2c0d8dd7bdc698dd1d58bde7651b682bcba4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341480
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41872}
This reverts commit aaa6851d53741179a591d79fc82c4dd6651a7ba5.
Reason for revert: breaks chromium webrtc import
Original change's description:
> Deprecate old constructors and set_type() in Candidate and Port
>
> * Deprecates constructors that use string based `type`
> * Deprecates string based type functions in favor of enum based.
> * Restrict possible values of Candidate::type. Ensure a valid value
> is assigned at construction.
> * Make Port constructors protected to limit their use to subclasses.
> - The reason for this is to make sure that use of SharedSocket()
> is controlled (it adds a bit of complexity).
> * Simplify construction of Port (remove Construct() etc)
>
> Bug: webrtc:15846
> Change-Id: If24ed674e175642efa49da37fd2bc847dd14f613
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339860
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41865}
Bug: webrtc:15846
Change-Id: Ic8b7cba97f8fb207ef51a88900e704658ade28b7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342140
Auto-Submit: Ilya Nikolaevskiy <ilnik@webrtc.org>
Owners-Override: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#41867}
* Deprecates constructors that use string based `type`
* Deprecates string based type functions in favor of enum based.
* Restrict possible values of Candidate::type. Ensure a valid value
is assigned at construction.
* Make Port constructors protected to limit their use to subclasses.
- The reason for this is to make sure that use of SharedSocket()
is controlled (it adds a bit of complexity).
* Simplify construction of Port (remove Construct() etc)
Bug: webrtc:15846
Change-Id: If24ed674e175642efa49da37fd2bc847dd14f613
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339860
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41865}
A call to GetScalabilityMode was added for logging purpose and causes an expectation failure for tests using 4 temporal layers.
Plan is to remove the old GetScalabilityMode and keep only the one that returns an optional.
Change-Id: I0e37a496bb621d9754d6572ef5838b58193aa183
Bug: b/327381318
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341520
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#41838}
Left in target are just .cc files with .h files used externally.
Bug: webrtc:14775
Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41835}
describing video codecs with their parameters as static members of SdpVideoFormat:
static const SdpVideoFormat VP8();
static const SdpVideoFormat H264();
static const SdpVideoFormat VP9Profile0();
static const SdpVideoFormat VP9Profile1();
static const SdpVideoFormat VP9Profile2();
static const SdpVideoFormat VP9Profile3();
static const SdpVideoFormat AV1Profile0();
static const SdpVideoFormat AV1Profile1();
This removes the need to craft instances of these by hand.
BUG=webrtc:15703
Change-Id: I2171e08b48ec98f18424f53f3b5d6d148130532e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337441
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41833}
This reverts commit d99499abbae94793a02944a1f28f7015816447f5.
Reason for revert: Breaks downstream projects and I can also repro locally when running the rtc_unittest test target (it does however pass in isolation indicating test cleanup/setup needs to be fixed)
Original change's description:
> p2p: separate ICE tie breaker and foundation seed
>
> BUG=webrtc:14626
>
> Change-Id: I189a708192c9cef0b50c3fcbe798b30376d3b547
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338982
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41806}
Bug: webrtc:14626
Change-Id: If45f8a33395c562c9388b3d3748e8566efa87ecb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341081
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Christoffer Dewerin <jansson@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Owners-Override: Christoffer Dewerin <jansson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41812}
