This CL works around an "Explicit specialization after instantiation
error" when building with clang-cl and is_component_build=true (see
crbug.com/1018579). On top of that it uses "template instantiation
declarations/declarations" in order to avoid to instantiate the
template in clients code.
TBR: hbos@webrtc.org
Bug: webrtc:9419, chromium:1018579
Change-Id: I1b2862de678586afc81e8f7a407947322f8a06c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158795
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29683}
I gave up on removing proxy_info, user_agent and tcp_options. I don't
think it's feasible to remove them without removing all the proxy code.
The assumption that you can set the proxy and user agent long after
you have created the factory is entrenched in unit tests and the code
itself. So is the ability to set tcp opts depending on protocol or
endpoint properties.
It may be easier to untangle proxy stuff from the factory later,
when it becomes a more first-class citizen and isn't passed via
the allocator.
Requires https://chromium-review.googlesource.com/c/chromium/src/+/1778870
to land first.
Bug: webrtc:7447
Change-Id: Ib496e2bb689ea415e9f8ec1dfedff13a83fa4a8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150799
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29091}
DataChannelTransportInterface takes the OpenChannel, SendData,
CloseChannel, and SetDataSink methods. MediaTransportInterface inherits
from DataChannelTransportInterface.
DatagramTransportInterface, the newer alternative to
MediaTransportInterface, also inherits from
DataChannelTransportInterface.
This will allow further refactors to enable the use of media-transport
style data channels alongside the datagram transport.
Bug: webrtc:9719
Change-Id: I2dd873785ea52d38055b62545c17e9e17c4e70c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147840
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28846}
This reverts commit 7f1c58938db72b1508e383d94a0e59dd70ff306e.
Reason for revert: this has been temporarily postponed.
Original change's description:
> Adding new top-level directory crypto/
>
> Adding the crypto root directory to WebRTC. The goal with this change is to
> centralize the management of crypto code into a single location.
>
> Currently we have cryptography code scattered across pc/ and rtc_base/
> which makes it difficult audit and maintain.
>
> By having a crypto/ directory we gain:
> 1. A clear first point of contact for auditing the cryptography in WebRTC.
> 2. Fine grain ownership to cryptography maintainers, we can include BoringSSL
> maintainers in this directory.
> 3. It improves maintanability of crypto code as we have improved modularization.
> It will not be deeply nested in all different parts of WebRTC.
> 4. Improved testability. We can cleanly build crypto libraries which plug into
> pc/ which we can more easily mock.
> 5. Enforce stricter rules. For example we may want to enforce ZeroOnFreeBuffer
> for all sensitive material. This is easier to enforce in a single directory.
>
> Bug: webrtc:9600
> Change-Id: I8e76332c7dcdac0a45a470ba2e930196e1ccf395
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125142
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27028}
TBR=steveanton@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,benwright@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9600
Change-Id: I3c99e733d53d76071179f0ff9ffdec965d20829d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147871
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28750}
This removes PeerConnectionInterface::SetBitrateAllocationStrategy()
plus a ton of now-dead code.
Bug: webrtc:10556
Change-Id: Icfae3bdd011588552934d9db4df16000847db7c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133169
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28523}
Adding the crypto root directory to WebRTC. The goal with this change is to
centralize the management of crypto code into a single location.
Currently we have cryptography code scattered across pc/ and rtc_base/
which makes it difficult audit and maintain.
By having a crypto/ directory we gain:
1. A clear first point of contact for auditing the cryptography in WebRTC.
2. Fine grain ownership to cryptography maintainers, we can include BoringSSL
maintainers in this directory.
3. It improves maintanability of crypto code as we have improved modularization.
It will not be deeply nested in all different parts of WebRTC.
4. Improved testability. We can cleanly build crypto libraries which plug into
pc/ which we can more easily mock.
5. Enforce stricter rules. For example we may want to enforce ZeroOnFreeBuffer
for all sensitive material. This is easier to enforce in a single directory.
Bug: webrtc:9600
Change-Id: I8e76332c7dcdac0a45a470ba2e930196e1ccf395
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125142
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27028}
This prepares for upcoming CL using the value for more than
controlling pacing rates.
Bug: webrtc:9887
Change-Id: Id3891c3727865149b87f946b3e7c3095a6ac9f26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126001
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27004}
While passing negative delta is an error it is not fatal and recovered next line.
Bug: None
Change-Id: I3b9ce234a7763ba92bd158c9eda8ba4bd7a06f4b
Reviewed-on: https://webrtc-review.googlesource.com/c/124702
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26916}
This involves inserting an extra layer between jsep_transport_controller
and the cricket::SctpTransportInternal layer. The objects at this layer
are reference counted.
Bug: chromium:818643
Change-Id: Ibed57c4a538de981cee63e0f7f1f319f029cab39
Reviewed-on: https://webrtc-review.googlesource.com/c/123884
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26889}
Hard-coding default values forces IDs over 14 to be used even
when we offer less than 15 different extensions.
Note that the code relies on MergeRtpHdrExts for making sure
that extension IDs are kept consistent and non-colliding between
different streams (audio/video).
Bug: webrtc:10288
Change-Id: I3e59f7ddc8ca43cea91084a6b7f36df70fb6be4a
Reviewed-on: https://webrtc-review.googlesource.com/c/121646
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26622}
Adds methods AddNetworkChangeCallback and RemoveNetworkChangeCallback,
to replace SetNetworkChangeCallback. Needed because both VideoChannel and
VoiceChannel register such a callback.
Bug: webrtc:9719
Change-Id: Ic592b2d775d721a0f44ba0af88ed963bf02d73a3
Reviewed-on: https://webrtc-review.googlesource.com/c/121460
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26575}
This creates the API for an ICE transport object, and lets it
be accessible from a DTLS transport object.
Bug: chromium:907849
Change-Id: Ieb24570217dec75ce0deca8420739c1f116fbba4
Reviewed-on: https://webrtc-review.googlesource.com/c/118703
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26472}
This is a followup, deleting the things marked as deprecated in
https://webrtc-review.googlesource.com/c/113180.
Bug: webrtc:9719
Change-Id: I64dc31c6918f575599fc6b0bbfa47c5b1f2f3019
Reviewed-on: https://webrtc-review.googlesource.com/c/113521
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26465}
The type rtc::scoped_refptr<T> is now part of api/. Please include it from
api/scoped_refptr.h.
More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o.
Bug: webrtc:9887, webrtc:8205
No-Try: True
Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf
Reviewed-on: https://webrtc-review.googlesource.com/c/119041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26414}
The flag rtc_use_builtin_sw_codecs will be removed in a later CL and
this marks usage of the various entry points using the old video factory
API as deprecated.
Bug: webrtc:7925, webrtc:10044
Change-Id: I5c75516a41b0666e77539c028808cc2b173ed4bd
Reviewed-on: https://webrtc-review.googlesource.com/c/113061
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25970}
Replaces enum VideoCodecType for video frames and uint8_t for audio
frames.
Also delete method
MediaTransportVideoSinkInterface::OnKeyFrameRequested; it needs to be
added as a send-side interface instead (for a later cl).
Bug: webrtc:9719
Change-Id: I2cfdbacc267afc75c448512e2cc6de0ec9966a2d
Reviewed-on: https://webrtc-review.googlesource.com/c/113180
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25918}
This includes a refactoring of jseptransport to store a refcounted
object instead of a std::unique_ptr to the cricket::DtlsTransport.
Bug: chromium:907849
Change-Id: Ib557ce72c2e6ce8af297c2b8deb7ec3a103d6d31
Reviewed-on: https://webrtc-review.googlesource.com/c/111920
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25831}
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/106280.
This time the whole code base is covered.
Some files may have not been fixed though, whenever the IWYU tool
was breaking the build.
Bug: webrtc:8311
Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef
Reviewed-on: https://webrtc-review.googlesource.com/c/111965
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25830}
This CL moves webrtc::CreatePeerConnectionFactory definitions out of
pc:create_pc_factory and merges it with its declaration in the api/
directory.
In order to avoid circular dependencies a new build target is created:
* api:create_peerconnection_factory
Bug: webrtc:9862
Change-Id: Ie215c94460cba026f5bf7d11c9a5aa03792064af
Reviewed-on: https://webrtc-review.googlesource.com/c/111186
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25744}
This reduces code duplication and ensures common behavior
between the unit classes.
Bug: webrtc:9709
Change-Id: I9529ef10b3f538355f53250a2b67c6b4e250cce8
Reviewed-on: https://webrtc-review.googlesource.com/c/110901
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25690}
Adds the types and methods required for sending and receiving data
channel messages over the media transport. These are:
- A DataMessageType to distinguish between text, binary, and control
messages
- A parameters struct for sending data messages, which specifies the
channel id, type, and ordering/reliability parameters
- A sink for data-channel related callbacks (receive data, begin
closing procedure, and end closing procedure)
- A method to set the sink for data channels
- Methods to open, close, and send on data channels
These methods, combined with the state sink, allow PeerConnection to
implement the DataChannelProviderInterface using MediaTransport as the
underlying transport.
Change-Id: Iccb2ba374594762a5b4f995564e2a1ff7d8805f5
Bug: webrtc:9719
Reviewed-on: https://webrtc-review.googlesource.com/c/108541
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25454}
After the problem fixed by [1], it seems wiser to only check that
`include_rules` contains all the top level directories and delegate the
action of adding a rule for .cc file to the first user of an header
from a new root level directory.
This should make the presubmit check more actionable (in [1] for example
the solution was to *not* add "+ios" but to consider ios/ as a non
WebRTC directory).
[1] - https://webrtc-review.googlesource.com/c/107707
Bug: webrtc:9887, webrtc:9924
Change-Id: Ic85e2153a2b83a4874c8faec3c5d1a8c61fe6faf
Reviewed-on: https://webrtc-review.googlesource.com/c/107731
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25386}
This CL updates the checkdeps configuration for the api/ folder in
order to explicitly avoid to #include non API headers from API headers.
In order to force a careful review of potential exceptions to this
rule, the CL also adds mbonadei@ and kwiberg@ as OWNERS of api/DEPS.
Bug: webrtc:9887
Change-Id: I0ada6f1020186b2782c7d060af36079c452ba1aa
Reviewed-on: https://webrtc-review.googlesource.com/c/106800
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25338}
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}
These files are required when implementing tests based on the test fixture,
and should be exposed as part of the test api.
This CL also removes a usage of stringstream and fixes some chromium-style
lint issues.
Bug: webrtc:8982, webrtc:163
Change-Id: I132aea0da79a79587887f21897236fc9802b7574
Reviewed-on: https://webrtc-review.googlesource.com/74586
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23346}
This moves them from an API directory (system_wrappers/include/) to a
non-API directory, which is exactly what we want for utilities like
this.
BUG=webrtc:8445
Change-Id: I6dc34fe662f5d87b3b5288d33055345bc6bf91db
Reviewed-on: https://webrtc-review.googlesource.com/21164
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22567}
The interface of the audioproc_f tool should be located in the api/ directory, so it becomes visible to the outside world.
Bug: webrtc:8732
Change-Id: Ia7475883aeb0e1f7a6afa5e791204b38dc53a8b8
Reviewed-on: https://webrtc-review.googlesource.com/61801
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22449}
The AEC3 factory is now part of the WebRTC API.
Bug: webrtc:8844
Change-Id: If6f419b4ca0354e2d346c0e6474086e456ba747e
Reviewed-on: https://webrtc-review.googlesource.com/57141
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22204}
We want api/peerconnectioninterface.h (and corresponding build target)
to not depend on call.h, and generally we treat Call as an internal,
non-api, class. But we need CallFactoryInterface in the api in order to
enable use of PeerConnection with or without support for media.
Making CallConfig a top-level class makes it possible to forward declare
it, together with Call, for use in callfactoryinterface.h and
peerconnectioninterface.h.
Delete the peerconnection_and_implicit_call_api target, replaced by
new target callfactory_api, to link between Call and Peerconnection.
Bug: webrtc:7504
Change-Id: I5e3978ef89bcd6705e94536f8676bcf89fc82fe1
Reviewed-on: https://webrtc-review.googlesource.com/46201
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22020}
- Move files from voice_engine/ to audio/.
- Rename voice_engine/utility.* to remix_resample.* since there are no other
utilities in those files.
- Move test/mock_voe_channel_proxy.h to audio/.
- Removed voe_channel_id from Audio[Receive|Send]Stream::Config.
- Remove VoiceEngine* from AudioState::Config.
- Fix a few cpplint complaints which showed when moving files.
NOPRESUBMIT=true
Bug: webrtc:4690
Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8
Reviewed-on: https://webrtc-review.googlesource.com/39268
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21657}
The new interface uses optionals instead of default values, and only values that are actually used are included. To make it easy to add/remove stats in the future, the struct itself is copied around on all layers, instead of copying the values one by one. This CL also fixes a bug which caused several APM statistics to get stuck at a fixed level when there are no more receive streams (after some period where there were receive streams). Since APM doesn't know this happens, an argument was added to the GetStats call to pass this information down to APM.
Bug: webrtc:8563, b/67926135
Change-Id: I96cc008353355bb520c4523f5c5379860f73ee24
Reviewed-on: https://webrtc-review.googlesource.com/25621
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20877}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
