OpenSSL implementations are all final implementations of their more abstract
SSL variants. This should be both documented and enforced by the use of the
final keyword to indicate to future WebRTC contributors that this is the
intended depth of inheritance and it shouldn't be extended again. Hopefully
this minor change will help keep the code simpler to maintain going forward.
Bug: webrtc:9860
Change-Id: Ie22de722214e3b209c3d7727a93ac819c112434e
Reviewed-on: https://webrtc-review.googlesource.com/c/108203
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25401}
This CL is the result of running include-what-you-use tool on part
of the code base (audio target and dependencies) plus manual fixes.
bug: webrtc:8311
Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604
Reviewed-on: https://webrtc-review.googlesource.com/c/106280
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25311}
There is no need to redefine SSL_CTX. base.h/ossl_typ.h defines it
already. Additionally, switch the base.h includes to the
OpenSSL-compatible ossl_typ.h spelling. That just got landed in
https://webrtc-review.googlesource.com/c/104120, so I'm guessing
OpenSSL consumers just didn't notice yet.
While getting the current BoringSSL name mangling scheme working with
WebRTC is a ways off, one of the requirements will almost certainly be
that WebRTC never forward-declare any BoringSSL types itself, instead
leaving it to openssl/base.h (or openssl/ossl_typ.h, the
OpenSSL-compatible alias). This is because we'd need to rename the
struct names themselves where they participate in C++ name mangling.
E.g. std::pair<RSA*, int> would mangle as rsa_st.
Bug: webrtc:5664
Change-Id: Ib9695d4ae4bc07d2bc54c9fdfb8600f44b5ec7bb
Reviewed-on: https://webrtc-review.googlesource.com/c/106675
Commit-Queue: David Benjamin <davidben@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25257}
This extends the API surface so that
custom certificates can be provided by an API user in both the standalone and
factory creation paths for the OpenSSLAdapter. Prior to this change the SSL
roots were hardcoded in a header file and directly included into
openssladapter.cc. This forces the 100 kilobytes of certificates to always be
compiled into the library. This is undesirable in certain linking cases where
these certificates can be shared from another binary that already has an
equivalent set of trusted roots hard coded into the binary.
Support for removing the hard coded SSL roots has also been added through a new
build flag. By default the hard coded SSL roots will be included and will be
used if no other trusted root certificates are provided.
The main goal of this CL is to reduce total binary size requirements of WebRTC
by about 100kb in certain applications where adding these certificates is
redundant.
Change-Id: Ifd36d92b5cb32d1b3098a61ddfc244d76df8f30f
Bug: chromium:526260
Change-Id: Ifd36d92b5cb32d1b3098a61ddfc244d76df8f30f
Reviewed-on: https://webrtc-review.googlesource.com/64841
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23180}
There was an implementation, but it relied on SSLCertificate::GetChain,
which was never implemented. Except in the fake certificate classes
used by the stats collector tests, hence the tests were passing.
Instead of implementing GetChain, we decided (in
https://webrtc-review.googlesource.com/c/src/+/6500) to add
methods that return a SSLCertChain directly, since it results in a
somewhat cleaner object model.
So this CL switches everything to use the "chain" methods, and gets
rid of the obsolete methods and member variables.
Bug: webrtc:8920
Change-Id: Ie9d7d53654ba859535462521b54c788adec7badf
Reviewed-on: https://webrtc-review.googlesource.com/56961
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22177}
Continue from https://codereview.webrtc.org/3010363002/
BUG=webrtc:8289
Change-Id: I8d14ba7974b654387f63ff30dee822b3045edcc7
Reviewed-on: https://webrtc-review.googlesource.com/6500
Commit-Queue: Jian Cui <jiancui@google.com>
Reviewed-by: David Benjamin <davidben@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20723}
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}
2017-09-15 04:25:06 +00:00
Renamed from webrtc/rtc_base/opensslidentity.h (Browse further)
