Bug: None
Change-Id: I1dd61f3363ba41ba94aa604ceac64b140fc72caa
Reviewed-on: https://webrtc-review.googlesource.com/61142
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22407}
So that we can avoid dependency cycles.
Bug: none
Change-Id: I821d9f1319dff01403d6e4e310cbb2d4b2b125e8
Reviewed-on: https://webrtc-review.googlesource.com/60500
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22328}
JsepTransportController process the entire SDP and handle the RTCP-mux,
SRTP setup, BUNDLE related logic internally. This will replace the current
TransportController.
JsepTransport2 is used by the JsepTransportController which processes the
transport part of SDP and owns the DtlsTransport created internally.
JsepTransport2 will replace JsepTransport and be renamed eventually.
Bug: webrtc:8587
Change-Id: Ib02dfa52fe9b7a5b8b132afcc8e4363eb8bd9cf4
Reviewed-on: https://webrtc-review.googlesource.com/48841
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22164}
This change list passes the instance of RtcEventLog from Peerconnection
down to P2PTransportChannel, and binds the structured ICE logging with
ICE layer objects. Logs of ICE connectivity checks are injected for
candidate pairs.
TBR=terelius@webrtc.org
Bug: None
Change-Id: Ia979dbbac6d31dcf0f8988da1065bdfc3e461821
Reviewed-on: https://webrtc-review.googlesource.com/34660
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21884}
If a WebRTC build target requires gmock it has to include
test/gmock.h and just depend on //test:test_support.
Unfortunately //testtest_support was a leaky abstraction because it
wasn't propagating the correct -I compiler flag. To make everything
work, all the targets that use gmock started also to depend on
//testing/gmock (even if they were not including any gmock header
directly).
This CL makes //testtest_support propagate the include path up in the
dependency chain so it is possible to remove unused dependencies.
Note: all_dependent_configs should probably be used in the original
gmock target. There is an ongoing discussion about it. This CL solves
the problem on WebRTC side and it is forward compatible.
TBR=phoglund@webrtc.org
Bug: webrtc:8603
Change-Id: If08daf2ce9a6431a6e881a236743b4ec33b59ea7
Reviewed-on: https://webrtc-review.googlesource.com/44340
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21776}
This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613.
Reason for revert:
The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660
Original change's description:
> Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""
>
> This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16.
>
> Reason for revert:
>
> Breaks Chrome FYI:
>
> /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
> -> returned 1
> ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
> static_library(target_name) {
> ^----------------------------
> The item //content/renderer:renderer
> can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
> because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
> //third_party/webrtc/*
> //third_party/webrtc_overrides/*
> ]
>
> https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout
>
> Original change's description:
> > GN rtc_* templates: Set default visibility to webrtc_root + "/*"
> >
> > This means that by default, targets are visible to everything under
> > the WebRTC root, but not visible to anything else.
> >
> > API targets are manually tagged with visibility "*", so that targets
> > outside the WebRTC tree can see them.
> >
> > BUG=webrtc:8254
> >
> > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> > Reviewed-on: https://webrtc-review.googlesource.com/24140
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21548}
>
> TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
>
> Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8254
> Reviewed-on: https://webrtc-review.googlesource.com/38760
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21555}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org
Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38860
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21558}
This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16.
Reason for revert:
Breaks Chrome FYI:
/b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
-> returned 1
ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
static_library(target_name) {
^----------------------------
The item //content/renderer:renderer
can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
//third_party/webrtc/*
//third_party/webrtc_overrides/*
]
https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout
Original change's description:
> GN rtc_* templates: Set default visibility to webrtc_root + "/*"
>
> This means that by default, targets are visible to everything under
> the WebRTC root, but not visible to anything else.
>
> API targets are manually tagged with visibility "*", so that targets
> outside the WebRTC tree can see them.
>
> BUG=webrtc:8254
>
> Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> Reviewed-on: https://webrtc-review.googlesource.com/24140
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21548}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38760
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21555}
This means that by default, targets are visible to everything under
the WebRTC root, but not visible to anything else.
API targets are manually tagged with visibility "*", so that targets
outside the WebRTC tree can see them.
BUG=webrtc:8254
Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
Reviewed-on: https://webrtc-review.googlesource.com/24140
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21548}
Was just being kept around until downstream dependents were updated
to include other files.
BUG=None
Change-Id: I26f0bea83c026be8a50a259c229ef8e7445825ad
Reviewed-on: https://webrtc-review.googlesource.com/35009
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21395}
SDP is a detail of PeerConnection and is not used by anything in p2p, so
it belongs in the pc/ directory. This also allows
MediaContentDescription to be inlined in the future.
Bug: webrtc:8620
Change-Id: I38b65ede9942e29eb15035ab29f2be988da1e5ce
Reviewed-on: https://webrtc-review.googlesource.com/33781
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21376}
This is a reland of 4770fd935ac92400487bddd3b755753572e6d692
Original change's description:
> Move JsepTransport from p2p/base to pc/.
>
> The JsepTransport class is moved to pc/ and the utility methods and
> enums are moved to where they are used.
>
> With JsepTransport moved to pc/, JsepTransport can depend on objects in
> pc/ including RtpTranport, SrtpTransport etc.
>
> Forked from https://webrtc-review.googlesource.com/c/src/+/31762/7
>
> Bug: webrtc:8636
> Change-Id: I4e8569fe3012946e87deb280f6139f0fd98de34d
> Reviewed-on: https://webrtc-review.googlesource.com/33701
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21333}
Bug: webrtc:8636
Change-Id: Ibce42be898b96dd8e0266b595611d2ffc86581a8
Reviewed-on: https://webrtc-review.googlesource.com/34586
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21371}
https://webrtc.googlesource.com/src.git/+/26246cac660a95f439b7d1c593edec2929806d3f
that was reverted due to compile error on windows.
Changes since last is an addition of a cast to uint16_t in stun.cc:1018.
---
Add RelayPortFactoryInterface that allows for custom relay (e.g turn) ports
This patch adds a RelayPortFactoryInterface that allows
for custom relay ports. The factor is added as optional argument
to BasicPortAlloctor. If none is provided a default implementation
that mimics existing behavior is created.
The patch also adds 2 stun functions, namely to copy a
StunAttribute and to remove StunAttribute's from a StunMessage.
Bug: webrtc:8640
Change-Id: If23638317130060286f576c94401de55c60a1821
Reviewed-on: https://webrtc-review.googlesource.com/34181
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21345}
This reverts commit 4770fd935ac92400487bddd3b755753572e6d692.
Reason for revert: breaks downstream projects
Original change's description:
> Move JsepTransport from p2p/base to pc/.
>
> The JsepTransport class is moved to pc/ and the utility methods and
> enums are moved to where they are used.
>
> With JsepTransport moved to pc/, JsepTransport can depend on objects in
> pc/ including RtpTranport, SrtpTransport etc.
>
> Forked from https://webrtc-review.googlesource.com/c/src/+/31762/7
>
> Bug: webrtc:8636
> Change-Id: I4e8569fe3012946e87deb280f6139f0fd98de34d
> Reviewed-on: https://webrtc-review.googlesource.com/33701
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21333}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org,pthatcher@webrtc.org
Change-Id: Ia72c6d7f185a95b21fd0aec90e7fdc00cb1fb423
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8636
Reviewed-on: https://webrtc-review.googlesource.com/34600
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21335}
The JsepTransport class is moved to pc/ and the utility methods and
enums are moved to where they are used.
With JsepTransport moved to pc/, JsepTransport can depend on objects in
pc/ including RtpTranport, SrtpTransport etc.
Forked from https://webrtc-review.googlesource.com/c/src/+/31762/7
Bug: webrtc:8636
Change-Id: I4e8569fe3012946e87deb280f6139f0fd98de34d
Reviewed-on: https://webrtc-review.googlesource.com/33701
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21333}
This reverts commit 26246cac660a95f439b7d1c593edec2929806d3f.
Reason for revert: Introduces compile failure on MSVC, which is preventing rolls into Chromium.
Sample errors:
[12263/40346] CXX obj/third_party/webrtc/p2p/rtc_p2p/stun.obj
FAILED: obj/third_party/webrtc/p2p/rtc_p2p/stun.obj
ninja -t msvc -e environment.x64 -- E:\b\c\goma_client/gomacc.exe "e:\b\c\win_toolchain\vs_files\a9e1098bba66d2acccc377d5ee81265910f29272\vc\tools\msvc\14.11.25503\bin\hostx64\x64/cl.exe" /nologo /showIncludes @obj/third_party/webrtc/p2p/rtc_p2p/stun.obj.rsp /c ../../third_party/webrtc/p2p/base/stun.cc /Foobj/third_party/webrtc/p2p/rtc_p2p/stun.obj /Fd"obj/third_party/webrtc/p2p/rtc_p2p_cc.pdb"
../../third_party/webrtc/p2p/base/stun.cc(1018): error C2220: warning treated as error - no 'object' file generated
../../third_party/webrtc/p2p/base/stun.cc(1018): warning C4267: 'argument': conversion from 'size_t' to 'uint16_t', possible loss of data
Original change's description:
> Add RelayPortFactoryInterface that allows for custom relay (e.g turn) ports
>
> This patch adds a RelayPortFactoryInterface that allows
> for custom relay ports. The factor is added as optional argument
> to BasicPortAlloctor. If none is provided a default implementation
> that mimics existing behavior is created.
>
> The patch also adds 2 stun functions, namely to copy a
> StunAttribute and to remove StunAttribute's from a StunMessage.
>
> Bug: webrtc:8640
> Change-Id: I59bd51f0f5e2f8c187dff9fcf003a24c35ed037f
> Reviewed-on: https://webrtc-review.googlesource.com/32600
> Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21267}
TBR=jonaso@webrtc.org,pthatcher@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8640
Change-Id: Idf83a1111727d2b5188b9c123f7471be7e99e973
Reviewed-on: https://webrtc-review.googlesource.com/33600
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21304}
One reason for the circular deps is that common_types.h is a
historical dumping ground for various structs and defines that
are believed to be generally useful. I tried moving things out
that did not appear to be used downstream (StreamCounters,
RtpCounters etc) and moved the things that seemed used
(RtpHeader + supporting structs) to a new file api/rtp_headers.h.
This makes their place in the api more clear while moving out
the things that don't belong in the API in the first place.
I had to extract out typedefs.h from webrtc_common to resolve
another circular dependency. I believe checks includes typedefs,
but common depends on checks.
Bug: webrtc:7745
Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b
Reviewed-on: https://webrtc-review.googlesource.com/33001
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21295}
This patch adds a RelayPortFactoryInterface that allows
for custom relay ports. The factor is added as optional argument
to BasicPortAlloctor. If none is provided a default implementation
that mimics existing behavior is created.
The patch also adds 2 stun functions, namely to copy a
StunAttribute and to remove StunAttribute's from a StunMessage.
Bug: webrtc:8640
Change-Id: I59bd51f0f5e2f8c187dff9fcf003a24c35ed037f
Reviewed-on: https://webrtc-review.googlesource.com/32600
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21267}
This splits things out of rtc_base and makes dependencies explicit.
Bug: webrtc:6828
Change-Id: Id521896c3c43595349021c857bec216e429a0c8d
Reviewed-on: https://webrtc-review.googlesource.com/32780
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21264}
This splits things out of rtc_base and makes dependencies explicit.
Bug: webrtc:6828
Change-Id: Ib813c7bd9e4de7ab015acb917bc09ee7204ba7bd
Reviewed-on: https://webrtc-review.googlesource.com/31940
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21245}
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}
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 file has moved to api/candidate.h.
Bug: webrtc:7504
Change-Id: Ic008f6277b2c2256776e0da69b903842103b1c29
Reviewed-on: https://webrtc-review.googlesource.com/22002
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20643}
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}
This CL is the same CL we had at https://codereview.webrtc.org/3016513002/.
Since we cannot land it with Rietveld anymore let's move the discussion
to Gerrit.
BUG=webrtc:7646
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal
No-Try: True
Change-Id: I442660e6dc71612d6bbcf73764bd4c6f65fcb760
Reviewed-on: https://webrtc-review.googlesource.com/7982
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Cr-Commit-Position: refs/heads/master@{#20257}
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 b23ed7f1af467a228cbdc63e839cac8856e9df8d.
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}
The TransportController was in p2p/base before and it cannot depend on
pc/ or media/ level targets because of the circular dependency. To make the
TransportController be responsible for creating and managing
the RtpTransport related objects which are pc/ level targets, the
TransportController is moved from p2p/base to pc/.
The TransportController makes more sense in pc/ anyway, since its main
responsibility is processing the "transport" parts of SDP which is
PeerConnection-specific.
This is also easier than moving RtpTransport related objects to p2p/base
because those objects also depend on other media/ and pc/ level targets
such as srtpfilter, cryptoparams etc.
Bug: webrtc:7013
Change-Id: Ic48dd5c454046ff3c81331f4b459f96a3255f328
Reviewed-on: https://webrtc-review.googlesource.com/4560
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20049}
This reverts commit 57fb3154b5411934b80051ad827db4e54d00f381.
Reason for revert: Breaks jingle_glue in chromium; need to leave candidate.h in place and include the new location until it's fixed.
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}
TBR=phoglund@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,perkj@webrtc.org
Change-Id: Ic5c3d0cf0b8c4d48ecbc49efdb76b373e3c950a5
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7504
Reviewed-on: https://webrtc-review.googlesource.com/4702
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20036}
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}
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}
