We use Optional in our public API, so its header should be in
webrtc/api/.
BUG=webrtc:8205
Review-Url: https://codereview.webrtc.org/3011943002
Cr-Commit-Position: refs/heads/master@{#19693}
directives in our DEPS files are not needed anymore.
Includes from webrtc/rtc_base are also whitelisted in webrtc/DEPS
so we don't have to whitelist it in all the others DEPS files.
BUG=webrtc:7634
NOTRY=True
Review-Url: https://codereview.webrtc.org/3006583002
Cr-Commit-Position: refs/heads/master@{#19601}
[This CL is work in progress.]
Wire up the rtp keep-alive in webrtc::Call::Config using new
SetRtpTransportParameters() method on RtpTransportInterface.
BUG=webrtc:7907
Review-Url: https://codereview.webrtc.org/2981513002
Cr-Commit-Position: refs/heads/master@{#19287}
Added documentation of thread expectations for video tracks and sources to the API.
Originally landed as patchset #2 id:20001 of https://codereview.webrtc.org/2964863002/.
Patchset 1 is the originall cl.
Patschet 2 is modified so that VideoTrackInterface::AddSink and RemoveSink have a default implementation.
BUG=none
Review-Url: https://codereview.webrtc.org/2989113002
Cr-Commit-Position: refs/heads/master@{#19195}
All downstream code have been updated to the new location.
In PRESUBMIT.py:
* Remove webrtc/rtc_base from CPP_BLACKLIST
* Add webrtc/rtc_base to LEGACY_API_DIRS
Fix some duplicated paths in
webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
BUG=webrtc:7634
TBR=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2976293002
Cr-Commit-Position: refs/heads/master@{#19094}
All downstream code have been updated to the new location.
In PRESUBMIT.py:
* Remove webrtc/rtc_base from CPP_BLACKLIST
* Add webrtc/rtc_base to LEGACY_API_DIRS
Fix some duplicated paths in
webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
BUG=webrtc:7634
TBR=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2973183002
Cr-Commit-Position: refs/heads/master@{#18948}
Reason for revert:
It breaks a downstream project.
Original issue's description:
> Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on the worker thread.
>
> Added documentation of thread expectations for video tracks and sources to the API.
>
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2964863002
> Cr-Commit-Position: refs/heads/master@{#18938}
> Committed: f1377f7222TBR=deadbeef@webrtc.org,noahric@chromium.org,yujo@chromium.org,perkj@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=None
Review-Url: https://codereview.webrtc.org/2979493003
Cr-Commit-Position: refs/heads/master@{#18942}
Added documentation of thread expectations for video tracks and sources to the API.
BUG=None
Review-Url: https://codereview.webrtc.org/2964863002
Cr-Commit-Position: refs/heads/master@{#18938}
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`
BUG=webrtc:7634
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
Review-Url: https://codereview.webrtc.org/2969623003
Cr-Commit-Position: refs/heads/master@{#18870}
[This CL is a rebase of an original CL by solenberg@:
https://codereview.webrtc.org/2948763002/ which in turn was a
rebase of an original CL by peah@:
https://chromium-review.googlesource.com/c/527032/]
Allow an external audio processing module to be used in WebRTC
This CL adds support for optionally using an externally created audio
processing module in a peerconnection. The ownership is shared
between the peerconnection and the external creator of the module.
As part of this the internal ownership of the audio processing module
is moved from VoiceEngine to WebRtcVoiceEngine.
BUG=webrtc:7775
Review-Url: https://codereview.webrtc.org/2961723004
Cr-Commit-Position: refs/heads/master@{#18837}
I intended to do this originally, but just forgot to pass the thread
with the virtual socket server into OrtcFactory::Create...
BUG=None
TBR=pthatcher@webrtc.org
Review-Url: https://codereview.webrtc.org/2967453002
Cr-Commit-Position: refs/heads/master@{#18831}
It was used to force a codec to the top of a list, but it resulted in
"a < a" being true, which some C++ runtimes complain about.
BUG=None
TBR=pthatcher@webrtc.org
Review-Url: https://codereview.webrtc.org/2963543002
Cr-Commit-Position: refs/heads/master@{#18786}
The only thing the physical socket server was used for was
"Wait"/"WakeUp", but it could be replaced by a simple rtc::Event.
So, removing this dependency makes things less confusing; the fact that
VirtualSocketServer takes a PhysicalSocketServer may lead someone to
think it uses real sockets internally, when it doesn't.
BUG=None
Review-Url: https://codereview.webrtc.org/2883313003
Cr-Commit-Position: refs/heads/master@{#18172}
Instead, make the pointer to the associated socket server a
construction time const, and delete its lock.
Introduces a helper class AutoSocketServerThread for code
(mainly tests) which need a socket server associated with
the current thread.
BUG=webrtc:7501
Review-Url: https://codereview.webrtc.org/2828223002
Cr-Commit-Position: refs/heads/master@{#18047}
Plumbed AudioEncoderFactory up into CreatePeerConnectionFactory.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2799033006
Cr-Commit-Position: refs/heads/master@{#17977}
I disabled the check on "video_tests" because it pulls
"//webrtc/media/rtc_unittest_main" as a dependency and it defines
the _main (that is already defined by "//webrtc/test:test_main").
I will file a bug to solve this in another CL.
BUG=webrtc:6828
NOTRY=True
Review-Url: https://codereview.webrtc.org/2832063003
Cr-Commit-Position: refs/heads/master@{#17859}
These tests are already built into rtc_unittests, so they end up being
run three times. Fixed by creating a "p2p_test_utils" target that
contains the test utils that ortc_unittests and rtc_media_unittests
depend on, but not the tests themselves.
BUG=None
TBR=kjellander@webrtc.org
Review-Url: https://codereview.webrtc.org/2820263004
Cr-Commit-Position: refs/heads/master@{#17752}
Reason for revert:
Breaks checkdeps rules. Need to make a "p2p_test_utils" build target to include things like fakeicetransport.h.
Original issue's description:
> Remove rtc_p2p_unittests from ortc_unittests executable.
>
> These tests are already built into rtc_unittests; they shouldn't be
> built into two test executables.
>
> BUG=None
> TBR=kjellander@webrtc.org
> NOTRY=True
>
> Review-Url: https://codereview.webrtc.org/2820263004
> Cr-Commit-Position: refs/heads/master@{#17748}
> Committed: fe9d38f515TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None
Review-Url: https://codereview.webrtc.org/2826703002
Cr-Commit-Position: refs/heads/master@{#17749}
The function converts the cricket::Codec, cricket::RtpHeaderExtensions
and cricket::StreamParamsVec to webrtc::RtpParameters.
BUG=webrtc:7311
Review-Url: https://codereview.webrtc.org/2735853004
Cr-Commit-Position: refs/heads/master@{#17124}
Introduce new target //webrtc/p2p:rtc_p2p_test_utils to host
test-related utilities.
Previously uncovered header "base/fakecandidatepair.h" is now also in a target.
BUG=webrtc:6828
Review-Url: https://codereview.webrtc.org/2714263004
Cr-Commit-Position: refs/heads/master@{#17036}
Create the SrtpTransportInterface, a subclass of RtpTransportInterface, which
allows the user to set the send and receive keys. The functionalities are
implemented inside the RtpTransportAdapters on top of BaseChannel.
BUG=webrtc:7013
Review-Url: https://codereview.webrtc.org/2714813004
Cr-Commit-Position: refs/heads/master@{#17023}
This CL adds the following interfaces:
* RtpTransportController
* RtpTransport
* RtpSender
* RtpReceiver
They're implemented on top of the "BaseChannel" object, which is normally used
in a PeerConnection, and roughly corresponds to an SDP "m=" section. As a result
of this, there are several limitations:
* You can only have one of each type of sender and receiver (audio/video) on top
of the same transport controller.
* The sender/receiver with the same media type must use the same RTP transport.
* You can't change the transport after creating the sender or receiver.
* Some of the parameters aren't supported.
Later, these "adapter" objects will be gradually replaced by real objects that don't
have these limitations, as "BaseChannel", "MediaChannel" and related code is
restructured. In this CL, we essentially have:
ORTC adapter objects -> BaseChannel -> Media engine
PeerConnection -> BaseChannel -> Media engine
And later we hope to have simply:
PeerConnection -> "Real" ORTC objects -> Media engine
See the linked bug for more context.
BUG=webrtc:7013
TBR=stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/2675173003
Cr-Commit-Position: refs/heads/master@{#16842}
