on an IPv6 network that contains the actual default local address. This is for preventing potential IP leaking.
BUG=webrtc:5376
Review URL: https://codereview.webrtc.org/1837823005
Cr-Commit-Position: refs/heads/master@{#12417}
This is a new way of generating RTCCertificate objects that is meant
to replace DtlsIdentityStoreInterface and all of its implementations
(clean up work).
It is similar to the identity store in that it generates on the worker
thread and does callback on the signaling thread, but:
- It does not generate identities in the background that you did not
ask for (preemptive generation made more sense before certificates
were parameterized, not so much anymore, and ECDSA which will be most
common takes like <=2 ms to generate).
- As such this code is less complicated than the store's code.
- The API is different, it takes Optional<uint64_t> expires and it
returns RTCCertificates, not SSLIdentities.
- It supports a blocking version of GenerateCertificate that can be
called from any thread, necessary for Chrome which can generate
certificates before the signaling/worker threads have been
initialized as WebRTC-threads (Chrome can invoke this version on
the worker thread outside of WebRTC).
This CL does not remove the identity store, only adds the alternative.
Follow-up CLs will start using it, the store will be removed once it
is no longer used anywhere.
BUG=webrtc:5707, webrtc:5708
R=hta@webrtc.org, torbjorng@webrtc.org
Review URL: https://codereview.webrtc.org/1883813002 .
Cr-Commit-Position: refs/heads/master@{#12381}
With this CL, legacy OpenSSL should work again.
BUG=webrtc:5714
Review URL: https://codereview.webrtc.org/1868033005
Cr-Commit-Position: refs/heads/master@{#12300}
- First audio RTP packet sent / received
- First RTP packet of the first video frame sent / received
- Last RTP packet of the first video frame sent / received
These timestamps should make it easier to measure how fast the call
becomes established from the user's perspective.
Review URL: https://codereview.webrtc.org/1765443002
Cr-Commit-Position: refs/heads/master@{#12287}
Instead of using a raw pointer output parameter. This affects
SSLStreamAdapter::GetPeerCertificate
Transport::GetRemoteSSLCertificate
TransportChannel::GetRemoteSSLCertificate
TransportController::GetRemoteSSLCertificate
WebRtcSession::GetRemoteSSLCertificate
This is a good idea in general, but will also be very convenient when
scoped_ptr is gone, since unique_ptr doesn't have an .accept() method.
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1802013002
Cr-Commit-Position: refs/heads/master@{#12262}
* Remove all source exclusions since they make the file very hard to
read and heavily increases the risk for mistakes.
* Don't compile the openssl* sources if use_openssl==0.
* Move platform specific sources into conditional includes to make it
easier to verify a 1:1 mapping with BUILD.gn (since GN doesn't support
automatic detection of platform specific sources based on filenames).
* Add missing sources for the GN build.
* Reorder some blocks to make GYP vs GN mapping match.
BUG=webrtc:4256
R=perkj@webrtc.org, torbjorng@webrtc.org
Review URL: https://codereview.webrtc.org/1859803002 .
Cr-Commit-Position: refs/heads/master@{#12235}
- Posts to WebRTC thread instead of Send
- Sample buffers are returned on capture session queue instead of main queue
- Camera switch happens on captures session queue
BUG=webrtc:5679, webrtc:4212
Review URL: https://codereview.webrtc.org/1838933004
Cr-Commit-Position: refs/heads/master@{#12186}
so that the call knows which packet ids were sent on the previous candidate pair.
Note that packet_id is actually 16bits, so we can use -1 for values that are not set.
Also moved the tests for candidate pair changes to TestSelectConnectionBeforeNomination.
BUG=
R=deadbeef@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1842093002 .
Cr-Commit-Position: refs/heads/master@{#12184}
This allows the reader to reference data, thus avoiding unnecessary
allocations and memory copies.
BUG=webrtc:5155,webrtc:5670
Review URL: https://codereview.webrtc.org/1821083002
Cr-Commit-Position: refs/heads/master@{#12160}
It'll go away soon, when rtc::scoped_ptr becomes a type alias for
std::unique_ptr.
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1837693002
Cr-Commit-Position: refs/heads/master@{#12144}
Reason for revert:
We will make it possible to link to BoringSSL for WebRTC's usages of the crypto APIs and OpenSSL for other usages in the same binary. Once that is completed, we will reland this.
Original issue's description:
> Remove code interfacing legacy openssl.
>
> BUG=webrtc:5664
>
> Committed: https://crrev.com/4cd331beade6de16c073dcdaf89c4e038bdbf73f
> Cr-Commit-Position: refs/heads/master@{#12041}
TBR=tommi@webrtc.org,davidben@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5664
Review URL: https://codereview.webrtc.org/1828773003 .
Cr-Commit-Position: refs/heads/master@{#12117}
Also include that in the stun-ping request as part of the
network-info attribute.
Change the network cost to be 16 bits.
BUG=
Review URL: https://codereview.webrtc.org/1815473002
Cr-Commit-Position: refs/heads/master@{#12110}
We shouldn't make external users define this flag to use our file.
BUG=
Review URL: https://codereview.webrtc.org/1825713003
Cr-Commit-Position: refs/heads/master@{#12106}
rtc::ThreadManager::ThreadManager() calls rtc::InitCocoaMultiThreading()
on iOS so add base/maccocoathreadhelper.mm to the file to build on iOS.
Fixes the following linker error:
Undefined symbols for architecture x86_64:
"rtc::InitCocoaMultiThreading()", referenced from:
rtc::ThreadManager::ThreadManager() in librtc_base.a(thread.o)
BUG=459705
NOTRY=True
Review URL: https://codereview.webrtc.org/1810373003
Cr-Commit-Position: refs/heads/master@{#12085}
webrtc/base/base.gyp unconditionally set SSL_USE_OPENSSL and
HAVE_OPENSSL_SSL_H, fix webrtc/base/BUILD.gn to do the same.
Better implementation than https://codereview.webrtc.org/1441323002
to fix the same underlying issue (i.e. compilation on iOS).
BUG=459705
Review URL: https://codereview.webrtc.org/1812213002
Cr-Commit-Position: refs/heads/master@{#12078}
This CL removes copy and assign support from Buffer and changes various
parameters from Buffer to CopyOnWriteBuffer so they can be passed along
and copied without actually copying the underlying data.
With this changed some parameters to be "const" and fixed an issue when
creating a CopyOnWriteBuffer with empty data.
BUG=webrtc:5155
Review URL: https://codereview.webrtc.org/1823503002
Cr-Commit-Position: refs/heads/master@{#12062}
Reason for revert:
I'm really sorry for having to revert this but it seems this hit an unexpected compile error downstream:
webrtc/media/sctp/sctpdataengine.cc: In function 'void cricket::VerboseLogPacket(const void*, size_t, int)':
webrtc/media/sctp/sctpdataengine.cc:172:37: error: invalid conversion from 'const void*' to 'void*' [-fpermissive]
data, length, direction)) != NULL) {
^
In file included from webrtc/media/sctp/sctpdataengine.cc:20:0:
third_party/usrsctp/usrsctplib/usrsctp.h:964:1: error: initializing argument 1 of 'char* usrsctp_dumppacket(void*, size_t, int)' [-fpermissive]
usrsctp_dumppacket(void *, size_t, int);
^
I'm sure you can fix this easily and just re-land this CL, while I'm going to look into how to add this warning at the public bots (on Monday).
Original issue's description:
> Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.
>
> This CL removes copy and assign support from Buffer and changes various
> parameters from Buffer to CopyOnWriteBuffer so they can be passed along
> and copied without actually copying the underlying data.
>
> With this changed some parameters to be "const" and fixed an issue when
> creating a CopyOnWriteBuffer with empty data.
>
> BUG=webrtc:5155
>
> Committed: https://crrev.com/944c39006f1c52aee20919676002dac7a42b1c05
> Cr-Commit-Position: refs/heads/master@{#12058}
TBR=kwiberg@webrtc.org,tkchin@webrtc.org,tommi@webrtc.org,pthatcher@webrtc.org,jbauch@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5155
Review URL: https://codereview.webrtc.org/1817753003
Cr-Commit-Position: refs/heads/master@{#12060}
The other buffer classes as well as all other dependencies are in rtc_base_approved, so I think this is a better place for it. Additionally I found that code in Chromium that already depends on the other buffer classes but now depends on the CopyOnWriteBuffer class, needed to have their build files updated and they previously depended on the buffer classes in rtc_base_approved.
TBR=jbauch@webrtc.org
Review URL: https://codereview.webrtc.org/1820643002
Cr-Commit-Position: refs/heads/master@{#12059}
This CL removes copy and assign support from Buffer and changes various
parameters from Buffer to CopyOnWriteBuffer so they can be passed along
and copied without actually copying the underlying data.
With this changed some parameters to be "const" and fixed an issue when
creating a CopyOnWriteBuffer with empty data.
BUG=webrtc:5155
Review URL: https://codereview.webrtc.org/1785713005
Cr-Commit-Position: refs/heads/master@{#12058}
No operator== that accepts one unique_ptr<T> and one T*. No implicit
conversion to bool. No rtc_make_scoped_ptr function.
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1803833002
Cr-Commit-Position: refs/heads/master@{#12048}
Also changed from unsigned to signed integer per the style guide.
By the way, I kept all delta-times to be 32-bit int.
The only things left in the p2p dir are
1. proberprober/main.cc where Time() is used as the input for a random number.
2. pseudotcp.cc: where 32-bit time info is sent over the wire.
BUG=webrtc:5636
Review URL: https://codereview.webrtc.org/1793553002
Cr-Commit-Position: refs/heads/master@{#12019}
Reason for revert:
The openmax_dl include change breaks downstream projects.
Original issue's description:
> Add check_deps rules in DEPS files.
>
> Add fine-grained check_deps rules for all of WebRTC.
> This will help both maintaining sane dependencies and provides a way
> to visualize dependency graphs using the buildtools/checkdeps/graphdeps.py script.
>
> Example:
> buildtools/checkdeps/graphdeps.py --root=. --format=png \
> --out=./webrtc.png --incl='^webrtc/modules/bitrate_controller->' \
> --excl='chromium|base|external|testing|webrtc/test|\.h$|\.cc$'
>
> will produce a neat webrtc.png image showcasing the dependencies
> (according to the DEPS file) for the bitrate_controller module.
> Some dependencies are filtered out for readability.
>
> BUG=webrtc:5623
> TESTED=Passing runs using:
> buildtools/checkdeps/checkdeps.py --root=. talk
> buildtools/checkdeps/checkdeps.py --root=. webrtc
>
> R=tommi@webrtc.org
>
> Committed: https://crrev.com/086f851b7b9b4bcbd4fe507c3bf83b760bd7f4d9
> Cr-Commit-Position: refs/heads/master@{#12008}
TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5623
Review URL: https://codereview.webrtc.org/1808573002
Cr-Commit-Position: refs/heads/master@{#12009}
Add fine-grained check_deps rules for all of WebRTC.
This will help both maintaining sane dependencies and provides a way
to visualize dependency graphs using the buildtools/checkdeps/graphdeps.py script.
Example:
buildtools/checkdeps/graphdeps.py --root=. --format=png \
--out=./webrtc.png --incl='^webrtc/modules/bitrate_controller->' \
--excl='chromium|base|external|testing|webrtc/test|\.h$|\.cc$'
will produce a neat webrtc.png image showcasing the dependencies
(according to the DEPS file) for the bitrate_controller module.
Some dependencies are filtered out for readability.
BUG=webrtc:5623
TESTED=Passing runs using:
buildtools/checkdeps/checkdeps.py --root=. talk
buildtools/checkdeps/checkdeps.py --root=. webrtc
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1796413002 .
Cr-Commit-Position: refs/heads/master@{#12008}
Part 3 of refactor. Also:
- better weak pointer delegate storage + tests
- we now ignore route changes when we're interrupted
- fixed bug where preferred sample rate wasn't set if audio session
wasn't active
BUG=
Review URL: https://codereview.webrtc.org/1796983004
Cr-Commit-Position: refs/heads/master@{#12007}
