Fixes
..\..\third_party\webrtc/base/stringutils.h(295,49) : warning(clang): extra qualification on member "empty_str" [-Wmicrosoft]
No behavior change, but makes the code more standards-conformant.
BUG=chromium:505296
Review URL: https://codereview.webrtc.org/1228193002
Cr-Commit-Position: refs/heads/master@{#9562}
Note: Regarding the ICMP6_CLOSE_FUNC variable in winping.cc,
Icmp6CloseHandle does not exist, and IcmpCloseHandle is the correct way
to close an IPv6 handle. Therefore the existing code is correct to use
close_ on both types of connections and this variable is unnecessary.
BUG=505319
Review URL: https://codereview.webrtc.org/1231653003
Cr-Commit-Position: refs/heads/master@{#9555}
Clang warns if there are missing braces around a subobject
initializer. The most common idiom that triggers this is:
STRUCT s = {0};
if the first field of STRUCT is itself a struct. This can
be more simply written as:
STRUCT s = {};
which also prevents the warning from firing.
Other instances of the warning have been fixed by adding
braces where appropriate.
BUG=505297
TBR=jiayl@webrtc.org
Review URL: https://codereview.webrtc.org/1216353002
Cr-Commit-Position: refs/heads/master@{#9529}
The old OpenSSL threading hooks were removed in favor of the library knowing
about threads internally. Instead of CRYPTO_add, use FOO_up_ref wrappers that
don't require reaching into the type.
BUG=none
R=jiayl@webrtc.org, juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/54579004
Cr-Commit-Position: refs/heads/master@{#9324}
If srtp_create fails while adding streams, it deallocates the session
but doesn't clear the passed pointer which then could lead to a
double-free in the SrtpSession dtor.
The CL also adds locking for libsrtp initialization / shutdown.
BUG=4042
R=jiayl@webrtc.org, juberti@google.com, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47319004
Cr-Commit-Position: refs/heads/master@{#9300}
Add pylintrc file based on
https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc
bit tightened up quite a bit (the one in depot_tools is far
more relaxed).
Remove a few excluded directories from pylint check and fixed/
suppressed all warnings generated.
Add GN format check + formatted all GN files using 'gn format'.
Cleanup redundant rules in tools/PRESUBMIT.py
TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations.
Ran it again with a modification in webrtc/build/webrtc.gni, formatted
all the GN files and ran it again.
R=henrika@webrtc.org, phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50069004
Cr-Commit-Position: refs/heads/master@{#9274}
This is being done in preparation of moving base/logging.* to rtc_base_approved. base/stream.* has libjingle dependencies that webrtc can't use, so logging.* can't depend on streams. It does look like stream.* isn't used much, so cleaning that up as well as cleaning up usage of the actual stream support (now LogStream) in the logging code, is in order, but I'll leave that to another cl.
BUG=
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54529004
Cr-Commit-Position: refs/heads/master@{#9269}
I wanted to use Config::Get in Chromium code, but it triggered the following
warning:
../../third_party/webrtc/common.h:89:20: error: declaration requires an exit-time destructor [-Werror,-Wexit-time-destructors]
static const T def;
^
../../third_party/webrtc/common.h:110:10: note: in instantiation of function template specialization requested here
return default_value<T>();
^
I assume we don't hit this in webrtc because the warning is disabled.
This also switches to the RTC_ prefix from the deprecated LIBJINGLE_.
Needed due to this Chromium CL:
https://codereview.chromium.org/1148843004/R=andresp@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53459004
Cr-Commit-Position: refs/heads/master@{#9268}
The current use of rtc::FifoBuffer can lead to reading across DTLS packet
boundaries which could cause packets to not being processed correctly.
This CL introduces the new class rtc::BufferQueue and changes the
StreamInterfaceChannel to use it instead of the rtc::FifoBuffer.
BUG=chromium:447431
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/52509004
Cr-Commit-Position: refs/heads/master@{#9254}
Previously SHA1Transform() kept a static buffer. As result SHA1 was not
always computed correctly when running that code in parallel on multiple
threads. That was causing spurious messages about invalid Message
Integrity attribute when running some tests in chromoting.
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/57379004
Cr-Commit-Position: refs/heads/master@{#9238}
This CL adds an API to SSL stream adapters to set the maximum allowed
protocol version and with that implements support for DTLS 1.2.
With DTLS 1.2 the default cipher changes in the unittests as follows.
BoringSSL
TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA -> TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256
NSS
TLS_RSA_WITH_AES_128_CBC_SHA -> TLS_RSA_WITH_AES_128_GCM_SHA256
BUG=chromium:428343
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/50989004
Cr-Commit-Position: refs/heads/master@{#9232}
Enable the -Wformat-security and -Wformat warnings for talk/.
Remove *.def and *.h.pump files from webrtc/base/base.gyp since they're not supported by some tools.
BUG=4242
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49969004
Cr-Commit-Position: refs/heads/master@{#9191}
This reduces the types exported in webrtc proper, which can cause other
issues (since it doesn't generally use webrtc/base/basictypes.h).
basictypes.h integral types (e.g. uint8) have been replaced by the
stdint counterparts (e.g. uint8_t), which matches general webrtc style.
The include for common.h has been replaced by constructormagic.h, which
was the only part used.
BUG=
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50859004
Cr-Commit-Position: refs/heads/master@{#9181}
This CL makes two changes to rtc::Buffer that have had to wait for
Chromium's use of it to be modernized:
1. Change default return type of rtc::Buffer::data() from char* to
uint8_t*. uint8_t is a more natural type for bytes, and won't
accidentally convert to a string. (Chromium previously expected
the default return type to be char, which is why
rtc::Buffer::data() initially got char as default return type in
9478437f, but that's been fixed now.)
2. Stop accepting void* inputs in constructors and methods. While
this is convenient, it's also dangerous since any pointer type
will implicitly convert to void*.
(This was previously committed (9e1a6d7c) but had to be reverted
(cbf09274) because Chromium on Android wasn't quite ready for it).
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47109004
Cr-Commit-Position: refs/heads/master@{#9132}
This is the second try at this. The previous try (a8e285d1) had to be
reverted (bd67f66e) because Chromium still declared a dependency on
move.h.
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48319004
Cr-Commit-Position: refs/heads/master@{#9123}
This CL makes two changes to rtc::Buffer that have had to wait for
Chromium's use of it to be modernized:
1. Change default return type of rtc::Buffer::data() from char* to
uint8_t*. uint8_t is a more natural type for bytes, and won't
accidentally convert to a string. (Chromium previously expected
the default return type to be char, which is why
rtc::Buffer::data() initially got char as default return type in
9478437f, but that's been fixed now.)
2. Stop accepting void* inputs in constructors and methods. While
this is convenient, it's also dangerous since any pointer type
will implicitly convert to void*.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44269004
Cr-Commit-Position: refs/heads/master@{#9121}
Removes FixedSizeLockFreeQueue which isn't used anymore. This enabled
moving rtc::AtomicOps to webrtc/base/atomicops.h where they should be.
BUG=4330
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51789004
Cr-Commit-Position: refs/heads/master@{#9120}
This pattern (read-only base, writable subclass) was picked to maintain a *Buffer option that doesn't copy the source bits when parsing. ByteBuffer and Buffer both copy. I'm open to discussion on what the type relationship would be, though :)
Tests have been added to ensure the symmetric nature of read/write operations.
BUG=
R=bcornell@google.com, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45259005
Cr-Commit-Position: refs/heads/master@{#9107}
This class is being used from both libjingle and webrtc but we recently had a regression when we added dependency on libjingle's Thread class. This cleans that up and moves the implementation of CriticalSection and helper classes into the source file.
I'm also improving debugging facilities and constness.
BUG=
R=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51769004
Cr-Commit-Position: refs/heads/master@{#9089}
