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 ed8390d21a.
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 aaa6851d53.
>
> 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 aaa6851d53.
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 d99499abba.
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}
The feature isn't in use by Google and has proven to contain security
issues. It's time to remove it.
Bug: b/324864439
Change-Id: I80344eb2f2060469d2d69a54dc4519fdd02ab4ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340324
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41808}
Goal is to make PeerConnectionInterface methods pure virtual.
This is a split of cl https://webrtc-review.googlesource.com/c/src/+/340143 in order to be able to fix Chromium test RTCPeerConnectionHandlerTest.OnRenegotiationNeeded
Bug: none
Change-Id: I5eac4d9a96c1b594c9e2b3505ef2466046065dc8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340481
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41798}
To ensure CreateVideoDecoder is only used as a fallback when Create is not implemented,
and thus make it safer to migrate VideoDecoderFactory implementations to Create.
Bug: webrtc:15791
Change-Id: Ifb15cf1d303348949ba51a3bb4c91b855a06627f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339841
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41777}
Move code from P2PTransportChannel to Candidate, where we set the
foundation value for remote prflx candidates.
Bug: none
Change-Id: I7dbcb85bca35dca7297136b0706092dd8d2b153c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339902
Reviewed-by: Philipp Hancke <phancke@microsoft.com>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41774}
until the code duplication can be removed which requires breaking
up the circular dependency.
BUG=webrtc:15847
Change-Id: Icc5f27dfcda26b1fcf16b19f79005d8b52fb6af3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339903
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41771}
Switch to type_name() for things like logging since `type()` will
change to returning an enumeration value.
The functional change that this has is that log statements and
Connection::ToString() (used for logging) will contain "host"
instead of "local" and "srflx" instead of "stun".
Bug: webrtc:15846
Change-Id: I35c50d026e4578a25d51765d59c6f2e01b850c94
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339180
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41757}
Candidate::type() is currently how the name of the type is fetched,
but that getter returns a non-standard type name.
Instead, I'm adding a new getter, type_name(), will follow up with
updating dependent code that needs the string, to use type_name (and
adapt to potential dependency on "local" or "stun") and then switch
type() to be enum based.
Also adding a test file for Candidate with a couple of basic tests to
start with.
Bug: webrtc:15846
Change-Id: I9b78b2405a9f962a3c07eaa8e72a79854c6f5ceb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339660
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41740}
First commit in a series of commits to wire up the test output path utility that adds a random directory in the path, for problematic tests that run in concurrent execution environments.
Bug: webrtc:15833
Change-Id: I5e5b3940007be773d77dbbfc953efa810e4e3ea9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339522
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41738}
This is a reland of commit 050ffefd85
Original change's description:
> Extends WebRTC logs for software encoder fallback
>
> This CL extends logging related to HW->SW fallbacks on the encoder
> side in WebRTC. The goal is to make it easier to track down the
> different steps taken when setting up the video encoder and why/when
> HW encoding fails.
>
> Current logs are added on several lines which makes regexp searching
> difficult. This CL adds all related information on one line instead.
>
> Three new search tags are also added VSE (VideoStreamEncoder), VESFW
> (VideoEncoderSoftwareFallbackWrapper) and SEA (SimulcastEncoderAdapter). The idea is to allow searching for the tags to see correlated logs.
>
> It has been verified that these added logs also show up in WebRTC
> logs in Meet.
>
> Logs from the GPU process are not included due to the sandboxed
> nature which makes it much more complex to add to the native
> WebRTC log. I think that these simple logs will provide value as is.
>
> Example: https://gist.github.com/henrik-and/41946f7f0b10774241bd14d7687f770b
>
> Bug: b/322132132
> Change-Id: Iec58c9741a9dd6bab3236a88e9a6e45440f5d980
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339260
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41733}
NOTRY=true
Bug: b/322132132
Change-Id: I25dd34b9ba59ea8502e47b4c89cd111430636e08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339680
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41736}
This reverts commit 050ffefd85.
Reason for revert: Breaks downstream project.
Original change's description:
> Extends WebRTC logs for software encoder fallback
>
> This CL extends logging related to HW->SW fallbacks on the encoder
> side in WebRTC. The goal is to make it easier to track down the
> different steps taken when setting up the video encoder and why/when
> HW encoding fails.
>
> Current logs are added on several lines which makes regexp searching
> difficult. This CL adds all related information on one line instead.
>
> Three new search tags are also added VSE (VideoStreamEncoder), VESFW
> (VideoEncoderSoftwareFallbackWrapper) and SEA (SimulcastEncoderAdapter). The idea is to allow searching for the tags to see correlated logs.
>
> It has been verified that these added logs also show up in WebRTC
> logs in Meet.
>
> Logs from the GPU process are not included due to the sandboxed
> nature which makes it much more complex to add to the native
> WebRTC log. I think that these simple logs will provide value as is.
>
> Example: https://gist.github.com/henrik-and/41946f7f0b10774241bd14d7687f770b
>
> Bug: b/322132132
> Change-Id: Iec58c9741a9dd6bab3236a88e9a6e45440f5d980
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339260
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41733}
Bug: b/322132132
Change-Id: I24d0a4e71a43ac192485f1af208563a51d919865
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339661
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41735}
This CL extends logging related to HW->SW fallbacks on the encoder
side in WebRTC. The goal is to make it easier to track down the
different steps taken when setting up the video encoder and why/when
HW encoding fails.
Current logs are added on several lines which makes regexp searching
difficult. This CL adds all related information on one line instead.
Three new search tags are also added VSE (VideoStreamEncoder), VESFW
(VideoEncoderSoftwareFallbackWrapper) and SEA (SimulcastEncoderAdapter). The idea is to allow searching for the tags to see correlated logs.
It has been verified that these added logs also show up in WebRTC
logs in Meet.
Logs from the GPU process are not included due to the sandboxed
nature which makes it much more complex to add to the native
WebRTC log. I think that these simple logs will provide value as is.
Example: https://gist.github.com/henrik-and/41946f7f0b10774241bd14d7687f770b
Bug: b/322132132
Change-Id: Iec58c9741a9dd6bab3236a88e9a6e45440f5d980
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339260
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41733}
To reduce number calls to the CreateVideoDecoder
Bug: webrtc:15791
Change-Id: I5d6ecc2e5e68165d4e012b3ad7edb6eaa40e1913
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336420
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41706}
Rework transport_description_factory to only have non-DTLS mode for
testing, and rewrite tests accordingly.
Bug: webrtc:11066, chromium:804275
Change-Id: Ie7d477c4331c975e4e0a3034fbbb749ed9009446
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336880
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41697}
