Both of these features are in std::optional and the lack
of them is making Optional use in WebRTC more cumbersome.
We are currently looking at using a more fully-fledged library
for some of our standard utility classes. This is merely a
stop-gap measure.
Bug: None
Change-Id: I958a984fa97a42f6e407be1f38662553efeceac4
Reviewed-on: https://webrtc-review.googlesource.com/22920
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20688}
This will help keep ortc dependencies clean in the future, since
gn --check forces us to depend on components from which we include
headers.
cryptoparams.h moves into api/, but ortc appears to think it
should be there anyway. We could consider moving it into the ortc/ api,
but it doesn't appear to be specific to ortc.
Bug: webrtc:6828
Change-Id: Iddae438d10b5e84b2fbc52565364319e20f90613
Reviewed-on: https://webrtc-review.googlesource.com/22660
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20686}
Introduces the public API interface corresponding to the
standardized RtpTransceiver object in the WebRTC spec.
https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver
The RtpTransceiver will be the internal representation for both
Plan B and Unified Plan SDP, but the public API interface will
only support Unified Plan (existing users should continue to use
GetSenders/GetReceivers, which will still be supported).
Bug: webrtc:7600
Change-Id: I417ffda683209ba9a9b4cbd274f91ca8295779a7
Reviewed-on: https://webrtc-review.googlesource.com/21460
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20659}
Conditional visibility is complex to maintain and it is not well
supported by other build systems.
This CL removes it and falls back on the more relaxed visibility value
("*" in this case).
It is not a problem because the targets that are using conditional
visibility are all marked as "testonly" and this is probably enough to
keep the build graph clean.
Bug: None
Change-Id: I2d2b5ac9a02d08c2863950116db455976ee1459c
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/14902
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20658}
This means we can properly declare the dependency between
libjingle_peerconnection_api and video_frame_api. i420
pulls in system_wrappers, which can't be a dependency of
the public API.
Plan:
1) Land this CL + send out PSA
2) Make all direct users of i420_buffer depend on the
new video_frame_api_i420 target
3) Move i420_buffer.cc to the new target
4) Make libjingle_peerconnection_api depend on
video_frame_api, since it no longer contains i420 code
Bug: webrtc:7504
Change-Id: I30d90f2ac7af53748859bbde8aed92386d5501f9
Reviewed-on: https://webrtc-review.googlesource.com/9382
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20656}
hbos knows and makes changes to the webrtc-pc spec[1] and works on
making Chromium's RTCPeerConnection spec-compliant. This includes
knowing and interacting with WebRTC-layer PeerConnection/Interface and
sometimes making changes to it.
hbos would like to share the peerconnection* ownership responsibilty as
it is relevant and owning it will speed up some of the process.
[1] https://w3c.github.io/webrtc-pc/
Bug: None
NOTRY: True
Change-Id: I8f419b7fc6c7fcf19951aa3f304769c915300d1b
Reviewed-on: https://webrtc-review.googlesource.com/21327
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20649}
So that we don't have to be capable of creating one ourselves, which
requires a dependency on the audio codecs.
BUG=webrtc:8396
Change-Id: I5600da5e17f613b0e61a9fb0fbdb373fe42f855c
Reviewed-on: https://webrtc-review.googlesource.com/20220
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20641}
- Add alpha accessors to PlanarYuvBuffer interface, null by defualt.
- Add WrapI420ABuffer() that creates a container which implements these
accessors.
- Show the use via StereoDecoderAdapter.
This CL is the step 2 for adding alpha channel support over the wire in webrtc.
See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
CL that gives an idea about how it will come together.
Design Doc: https://goo.gl/sFeSUT
Bug: webrtc:7671
Change-Id: Id5691cde00088ec811b63d89080d33ad2d6e3939
Reviewed-on: https://webrtc-review.googlesource.com/21130
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20635}
Even if we're not going to transmit any timing info over the wire.
Bug: webrtc:8504
Change-Id: Id54192a10e6b2a6a2cb57a2ff6b28fc0d16e471d
Reviewed-on: https://webrtc-review.googlesource.com/21160
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20628}
We need to get rid of the ones that don't take audio codec factory
arguments in order to eliminate the dependency on audio codec
implementations.
BUG=webrtc:8396
Change-Id: Id0c1c3b70c2b3479da81ba1056cc69e857e454bd
Reviewed-on: https://webrtc-review.googlesource.com/12281
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20555}
Second attempt to land https://webrtc-review.googlesource.com/c/src/+/16180
Now removes voice_engine dependency from peerconnection and fixes a minor
const issue in NullAudioPoller.
TBR=solenberg
Bug: webrtc:7313
Change-Id: Ibfddbdc76118581e4a4dc64575203f84c1659e5c
Reviewed-on: https://webrtc-review.googlesource.com/17784
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20526}
This reverts commit 90bace0958.
Reason for revert: The original problem of this CL has been fixed in https://webrtc-review.googlesource.com/17540 but sounds like it is also adding voice_engine as a dependency of pc:peerconnection. We should investigate this because probably we can avoid it.
Original change's description:
> Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API
>
> (this CL is based on the work by Taylor and Steve in https://webrtc-review.googlesource.com/c/src/+/10201)
>
> This SetAudioPlayout method lets applications disable audio playout while
> still processing incoming audio data and generating statistics on the
> received audio.
>
> This may be useful if the application wants to set up media flows as
> soon as possible, but isn't ready to play audio yet. Currently, native
> applications don't have any API point to control this, unless they
> completely implement their own AudioDeviceModule.
>
> The SetAudioRecording works in a similar fashion but for the recorded
> audio. One difference is that calling SetAudioRecording(false) does not
> keep any audio processing alive.
>
> TBR=solenberg
>
> Bug: webrtc:7313
> Change-Id: I0aa075f6bfef9818f1080f85a8ff7842fb0750aa
> Reviewed-on: https://webrtc-review.googlesource.com/16180
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20499}
TBR=solenberg@webrtc.org,henrika@webrtc.org,kwiberg@webrtc.org
Change-Id: I8431227e21dbffcfed3dd0e6bd7ce26c4ce09394
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7313
Reviewed-on: https://webrtc-review.googlesource.com/17701
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20512}
(this CL is based on the work by Taylor and Steve in https://webrtc-review.googlesource.com/c/src/+/10201)
This SetAudioPlayout method lets applications disable audio playout while
still processing incoming audio data and generating statistics on the
received audio.
This may be useful if the application wants to set up media flows as
soon as possible, but isn't ready to play audio yet. Currently, native
applications don't have any API point to control this, unless they
completely implement their own AudioDeviceModule.
The SetAudioRecording works in a similar fashion but for the recorded
audio. One difference is that calling SetAudioRecording(false) does not
keep any audio processing alive.
TBR=solenberg
Bug: webrtc:7313
Change-Id: I0aa075f6bfef9818f1080f85a8ff7842fb0750aa
Reviewed-on: https://webrtc-review.googlesource.com/16180
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20499}
This CL is the step 1 for adding alpha channel support over the wire in webrtc.
- Add the footprint for adapter classes that wraps actual codecs.
- This CL does not add a webrtc::VideoFrame container that can carry alpha to
make the CL shorter for an easier review. Therefore, it exercises a code path
for when we receive no alpha input, just regular I420 frames.
- Unittest sends a video frame for encode/decode through these adapters and
checks the output PSNR.
- See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
CL that gives an idea about how it will come together.
Design Doc: https://goo.gl/sFeSUT
Bug: webrtc:7671
Change-Id: I9d3be13647a0a958feceb8d7a9aa93852fc6a1fa
Reviewed-on: https://webrtc-review.googlesource.com/11841
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20490}
We've done this previously with the other audio encoders, but Opus had
to wait until all external users had been updated.
BUG=webrtc:7847
Change-Id: I70422d7b6c715f32a43bee88febcf6b6155e18b3
Reviewed-on: https://webrtc-review.googlesource.com/8000
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20424}
Add fine grained dropped video frames counters on sending side
4 new counters added to SendStatisticsProxy and reported to UMA and logs.
Bug: webrtc:8355
Change-Id: I1f9bdfea9cbf17cf38b3cb2f55d406ffdb06614f
Reviewed-on: https://webrtc-review.googlesource.com/14580
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20421}
Originally, the idea was to implement QUIC data channels as a
PeerConnection API. Now, the effort has shifted to implementing it as a
part of ORTC which will live in Chromium. Since this code has not been
maintained and is not currently being used, remove it to reduce
maintenance overhead while a copy will be retained in the Git history.
Bug: webrtc:8385
Change-Id: I2719c007a0de0118b67d41a425f900b66c52f65a
Reviewed-on: https://webrtc-review.googlesource.com/14100
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20412}
4 new counters added to SendStatisticsProxy and reported to UMA and logs.
Bug: webrtc:8355
Change-Id: Idf9b8dfc295c92821e058a97cb3894dc6a446082
Reviewed-on: https://webrtc-review.googlesource.com/12260
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20347}
This reverts commit 54d1da13a5.
Reason for revert: Breaking tests
Original change's description:
> BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic.
>
> This CL implements the main logic and IOS appRTC integration.
>
> Unit tests and Android appRTC will be in separate CL.
>
> Bug: webrtc:8243
> Change-Id: If8e5195294046a47316e9fade1b0dfec211155e1
> Reviewed-on: https://webrtc-review.googlesource.com/4860
> Commit-Queue: Alex Narest <alexnarest@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20329}
TBR=deadbeef@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,alexnarest@webrtc.org
Change-Id: I5be1da78f360f72be66f9d56dd6b88c1cc13e963
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8243
Reviewed-on: https://webrtc-review.googlesource.com/12560
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20330}
This CL implements the main logic and IOS appRTC integration.
Unit tests and Android appRTC will be in separate CL.
Bug: webrtc:8243
Change-Id: If8e5195294046a47316e9fade1b0dfec211155e1
Reviewed-on: https://webrtc-review.googlesource.com/4860
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20329}
This will allow Chrome to provide a RtcEventLogOutput object that reports the log back to Chrome, allowing Chrome to manage the log by itself - write it to a file, upload it to a server, etc.
Bug: webrtc:8111
Change-Id: I6a2a6945fc8586ef10e0fb9c56eaa8fda00dfc98
Reviewed-on: https://webrtc-review.googlesource.com/8081
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20295}
This patch adds an interface that allows modification of stun messages
sent by TurnPort. A user can inject a TurnCustomizer on the RTCConfig
and the TurnCustomizer will be invoked by TurnPort before sending
message. This allows user to e.g add custom attributes as described
in rtf5389.
BUG=webrtc:8313
Change-Id: I6f4333e9f8ff7fd20f32677be19285f15e1180b6
Reviewed-on: https://webrtc-review.googlesource.com/7618
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20233}
This reverts commit 6c0c55c318.
Reason for revert:
Fixed the flake.
Original change's description:
> Revert "Added PeerConnectionObserver::OnRemoveTrack."
>
> This reverts commit ba97ba7af9.
>
> Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway.
>
> Original change's description:
> > Added PeerConnectionObserver::OnRemoveTrack.
> >
> > This corresponds to processing the removal of a remote track step of
> > the spec, with processing the addition of a remote track already
> > covered by OnAddTrack.
> > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks
> >
> > Bug: webrtc:8260, webrtc:8315
> > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700
> > Reviewed-on: https://webrtc-review.googlesource.com/4722
> > Commit-Queue: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20214}
>
> TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org
>
> Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8260, webrtc:8315
> Reviewed-on: https://webrtc-review.googlesource.com/7940
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Alex Loiko <aleloi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20218}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org
Change-Id: Iab7500bebf98535754b102874259de43831fff6b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8260, webrtc:8315
Reviewed-on: https://webrtc-review.googlesource.com/8180
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20227}
This will make it easier for users to specify that they want iSAC in
their codec factories, since they'll no longer have to worry about
choosing either the fix or the float implementation.
BUG=webrtc:8343
Change-Id: I5fb713710a8dd86162b5de73a2f0a851947f1411
Reviewed-on: https://webrtc-review.googlesource.com/6540
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20222}
This reverts commit ba97ba7af9.
Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway.
Original change's description:
> Added PeerConnectionObserver::OnRemoveTrack.
>
> This corresponds to processing the removal of a remote track step of
> the spec, with processing the addition of a remote track already
> covered by OnAddTrack.
> https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks
>
> Bug: webrtc:8260, webrtc:8315
> Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700
> Reviewed-on: https://webrtc-review.googlesource.com/4722
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20214}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org
Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8260, webrtc:8315
Reviewed-on: https://webrtc-review.googlesource.com/7940
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20218}
This corresponds to processing the removal of a remote track step of
the spec, with processing the addition of a remote track already
covered by OnAddTrack.
https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks
Bug: webrtc:8260, webrtc:8315
Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700
Reviewed-on: https://webrtc-review.googlesource.com/4722
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20214}
This reverts commit b23ed7f1af.
Reason for revert: Breaks Chromium FYI build
Sample error log:
../../remoting/test/fake_port_allocator.cc:52:7: error: no matching constructor for initialization of 'cricket::BasicPortAllocator'
: BasicPortAllocator(network_manager, socket_factory),
^ ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
../../third_party/webrtc/p2p/client/basicportallocator.h:32:12: note: candidate constructor not viable: requires single argument 'network_manager', but 2 arguments were provided
explicit BasicPortAllocator(rtc::NetworkManager* network_manager);
^
../../third_party/webrtc/p2p/client/basicportallocator.h:27:7: note: candidate constructor (the implicit copy constructor) not viable: requires 1 argument, but 2 were provided
class BasicPortAllocator : public PortAllocator {
^
../../third_party/webrtc/p2p/client/basicportallocator.h:29:3: note: candidate constructor not viable: requires 3 arguments, but 2 were provided
BasicPortAllocator(rtc::NetworkManager* network_manager,
^
../../third_party/webrtc/p2p/client/basicportallocator.h:33:3: note: candidate constructor not viable: requires 3 arguments, but 2 were provided
BasicPortAllocator(rtc::NetworkManager* network_manager,
^
../../third_party/webrtc/p2p/client/basicportallocator.h:36:3: note: candidate constructor not viable: requires 5 arguments, but 2 were provided
BasicPortAllocator(rtc::NetworkManager* network_manager,
Original change's description:
> TurnCustomizer - an interface for modifying stun messages sent by TurnPort
>
> This patch adds an interface that allows modification of stun messages
> sent by TurnPort. A user can inject a TurnCustomizer on the RTCConfig
> and the TurnCustomizer will be invoked by TurnPort before sending
> message. This allows user to e.g add custom attributes as described
> in rtf5389.
>
> BUG=webrtc:8313
>
> Change-Id: Ibf5cc10af84c57288f1eb4c578ca064611a769f1
> Reviewed-on: https://webrtc-review.googlesource.com/4781
> Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20197}
TBR=deadbeef@webrtc.org,sakal@webrtc.org,jonaso@webrtc.org
Change-Id: I624efb22f6e3ceac1b2ff8af1ec47e4cfdde9140
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8313
Reviewed-on: https://webrtc-review.googlesource.com/7680
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20199}
This patch adds an interface that allows modification of stun messages
sent by TurnPort. A user can inject a TurnCustomizer on the RTCConfig
and the TurnCustomizer will be invoked by TurnPort before sending
message. This allows user to e.g add custom attributes as described
in rtf5389.
BUG=webrtc:8313
Change-Id: Ibf5cc10af84c57288f1eb4c578ca064611a769f1
Reviewed-on: https://webrtc-review.googlesource.com/4781
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20197}
These versions of StartRtcEventLog() are not used.
Bug: webrtc:8111
Change-Id: I1fb543a908decff203b13f8358598f75d875c111
Reviewed-on: https://webrtc-review.googlesource.com/6782
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20191}
Move RtcEventLogOutput into the API, so that we would be able to change StartRtcEventLog (in PeerConnectionInterface) to use it.
Bug: webrtc:8111
Change-Id: I1d70af792ec584d3f1a8eced1b66c38e4a360642
Reviewed-on: https://webrtc-review.googlesource.com/7220
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20189}
This is in preparation for deleting the include in rtc_base/refcount.h,
but that change has to wait for some downstream applications to
not rely in the indirect include.
Partial reland of "Make rtc_base/refcount.h self contained, not including refcountedobject.h."
This is a reland of b7239a9dc8
Original change's description:
> Make rtc_base/refcount.h self contained, not including refcountedobject.h.
>
> The refcount.h file doesn't depend on anything from
> refcountedobject.h. The motivation of this change to make it possible
> to add additional declarations to refcount.h, and include it from
> refcountedobject.h.
>
> Bug: webrtc:8270
> Change-Id: I24f6131f471e675570968d00065ff9b1f55e3373
> Reviewed-on: https://webrtc-review.googlesource.com/5760
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20106}
Bug: webrtc:8270
Change-Id: I63a42712f6c1ec83823c629d1a954fd1a04d4a6c
Reviewed-on: https://webrtc-review.googlesource.com/7281
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20185}
This is a reland of b7239a9dc8
Original change's description:
> Make rtc_base/refcount.h self contained, not including refcountedobject.h.
>
> The refcount.h file doesn't depend on anything from
> refcountedobject.h. The motivation of this change to make it possible
> to add additional declarations to refcount.h, and include it from
> refcountedobject.h.
>
> Bug: webrtc:8270
> Change-Id: I24f6131f471e675570968d00065ff9b1f55e3373
> Reviewed-on: https://webrtc-review.googlesource.com/5760
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20106}
Bug: webrtc:8270
Change-Id: I9738f6680ab52d0f43639a1a39175fdba5957681
Reviewed-on: https://webrtc-review.googlesource.com/5840
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20180}
There are some internal projects that need to be updated before we remove StartRtcEventLog and StopRtcEventLog. In this CL we take away the pure-virtuality status of the functions. After landing this, we can fix the internal projects, then land https://webrtc-review.googlesource.com/c/src/+/6782.
TBR=stefan@webrtc.org
Bug: webrtc:8111
Change-Id: Ibe495a7e7d6bf8120b1a26f056bd1443031733bf
Reviewed-on: https://webrtc-review.googlesource.com/6980
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20178}
This is a reland of 9185aca9ce
> Original change's description:
> > > Clean up libjingle API dependencies.
> > >
> > > This CL moves candidate.h into the public API, since it has
> > > been implicitly included before.
> > >
> > > This is a straightforward way of solving the circular
> > > dependencies involving that file. For instance,
> > > libjingle_peerconnection_api includes candidate.h from
> > > jsepicecandidate.h, but _api can't depend on rtc_p2p, which
> > > depends on _api. In fact, _api can't depend on much at all
> > > since it's a very high level abstraction; instead, things
> > > should depend on it.
> > >
> > > Furthermore, we have the case where deprecated headers
> > > include headers in internal modules. I just have to turn
> > > off include checking for those, but that's not a big deal.
> > >
> > > This CL punts the problem of callfactoryinterface.h being
> > > implicitly included, and pulling in most of the call
> > > module with it. This should be addressed in a follow-up
> > > CL.
> Bug: webrtc:7504
> Change-Id: Icae0ba1a0488550e2871cc65e66d3661707aa5b6
> Reviewed-on: https://webrtc-review.googlesource.com/6460
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20156}
TBR=deadbeef@webrtc.org
Bug: webrtc:7504
Change-Id: Ic6598ac2af9355b60bbd289c86dc75e0ae9fed2e
Reviewed-on: https://webrtc-review.googlesource.com/6801
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20167}
This reverts commit 9185aca9ce.
Reason for revert: Still breaks Chromium:
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/29052
You cannot trust the *chromium* trybots due to https://bugs.chromium.org/p/chromium/issues/detail?id=771159
Original change's description:
> Reland "Clean up libjingle API dependencies."
>
> This is a reland of 5117b04787
> Original change's description:
> > > Clean up libjingle API dependencies.
> > >
> > > This CL moves candidate.h into the public API, since it has
> > > been implicitly included before.
> > >
> > > This is a straightforward way of solving the circular
> > > dependencies involving that file. For instance,
> > > libjingle_peerconnection_api includes candidate.h from
> > > jsepicecandidate.h, but _api can't depend on rtc_p2p, which
> > > depends on _api. In fact, _api can't depend on much at all
> > > since it's a very high level abstraction; instead, things
> > > should depend on it.
> > >
> > > Furthermore, we have the case where deprecated headers
> > > include headers in internal modules. I just have to turn
> > > off include checking for those, but that's not a big deal.
> > >
> > > This CL punts the problem of callfactoryinterface.h being
> > > implicitly included, and pulling in most of the call
> > > module with it. This should be addressed in a follow-up
> > > CL.
> > >
> > > Bug: webrtc:7504
> > > Change-Id: I1b1729408158418333ccdf702bf529386090f0d7
> > > Reviewed-on: https://webrtc-review.googlesource.com/2020
> > > Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
> > > Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#20034}
>
> Bug: webrtc:7504
> Change-Id: Icae0ba1a0488550e2871cc65e66d3661707aa5b6
> Reviewed-on: https://webrtc-review.googlesource.com/6460
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20156}
TBR=phoglund@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org
Change-Id: I699c68bd330b537005c3f2b8fe31702025df4e39
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7504
Reviewed-on: https://webrtc-review.googlesource.com/6800
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20157}
