Reason for revert:
Chromium build issues have been resolved.
Original issue's description:
> Revert of Remove all reference to carbon api (patchset #2 id:20001 of https://codereview.webrtc.org/2299633002/ )
>
> Reason for revert:
> Breaks chromium build
>
> Original issue's description:
> > Remove all reference to carbon api
> >
> > BUG=webrtc:6282
> >
> > Committed: https://crrev.com/dbd8b6bec4143c940b2f2ca8cd85c25d17327964
> > Cr-Commit-Position: refs/heads/master@{#14080}
>
> TBR=magjed@webrtc.org,mflodman@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6282
>
> Committed: https://crrev.com/b096aa7fd375a980daab3a986596548ca5de2a1c
> Cr-Commit-Position: refs/heads/master@{#14081}
TBR=magjed@webrtc.org,mflodman@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6282
Review-Url: https://codereview.webrtc.org/2321493002
Cr-Commit-Position: refs/heads/master@{#14125}
There is no clear reason to have them in build_overrides, and
webrtc/build seems to be a better place.
Also, delete build_overrides/webrtc.gni
NOTRY=True
BUG=webrtc:5949
Review-Url: https://codereview.webrtc.org/2309253004
Cr-Commit-Position: refs/heads/master@{#14108}
Reason for revert:
Breaks chromium build
Original issue's description:
> Remove all reference to carbon api
>
> BUG=webrtc:6282
>
> Committed: https://crrev.com/dbd8b6bec4143c940b2f2ca8cd85c25d17327964
> Cr-Commit-Position: refs/heads/master@{#14080}
TBR=magjed@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6282
Review-Url: https://codereview.webrtc.org/2316563002
Cr-Commit-Position: refs/heads/master@{#14081}
In addition to setting or appending from another Buffer, which was
already possible, this allows for e.g. std::vector and rtc::ArrayView
arguments.
Review-Url: https://codereview.webrtc.org/2293983002
Cr-Commit-Position: refs/heads/master@{#14073}
Remove common_inherited_config from the targets and add it to the
template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2311843002
Cr-Commit-Position: refs/heads/master@{#14069}
Before this change, the argument still needed those methods, but not
having them resulted in a compilation error. Now, it results in this
constructor being removed from the overload set.
This currently makes no difference, but I'm about to publish a CL that
breaks without this.
Review-Url: https://codereview.webrtc.org/2312473002
Cr-Commit-Position: refs/heads/master@{#14068}
Remove common_config from the targets' config and add
it to the template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2300413002
Cr-Commit-Position: refs/heads/master@{#14063}
Defines the rtc_executable, rtc_source_set, rtc_test and
rtc_static_library templates.
These templates provide no functionality yet, but will enable common
configuration to be introduced, avoiding repetition in every target
Changes summary:
- Prepend rtc_ to test, source_set, executable and static_library targets
- Change "configs -= [" to "suppressed_configs += ["
- Include webrtc/build/webrtc.gni where it wasn't included yet
- Delete import("//testing/test.gni"), since rtc_test makes it unnecessary.
BUG=webrtc:6187
TBR=henrik.lundin@webrtc.org,tommi@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2301053002
Cr-Commit-Position: refs/heads/master@{#14043}
The sample uses are from when I debugged bug 617124. The change in neteq_network_stats_unittest.cc is a fix for a minor unrelated bug found by the try bots when I tried to land this CL (a test was passing uninitialized packet data to NetEq).
BUG=chromium:617124
Review-Url: https://codereview.webrtc.org/2293893002
Cr-Commit-Position: refs/heads/master@{#14034}
WebRTC no longer has any restriction on what thread frames should be
delivered on. One possible problem with this CL is that NV21->I420
conversion and scaling is done on the thread that delivers frames, which
might cause fps regressions.
R=nisse@webrtc.org, perkj@webrtc.org, tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/2137503003 .
Cr-Commit-Position: refs/heads/master@{#14021}
This is step 2 of the plan below.
The modified plan
1. First land unmodified task_queue.h into webrtc_override in Chrome
2. Modify build files in the webrtc repo to include the task_queue.h and task_queue.cc from webrtc_overrides. This will breaks webrtc Chrome FYI.
3. Combine a roll of webrtc to Chrome and a the cl in https://codereview.chromium.org/2293913003/ into one cl.
The combined cl will roll in build files from 2 and add task_queue.cc in webrtc_overrides and build task_queue_unittest.cc as part of content_unittests to test task_queue.cc in webrtc_overrides.
4... Start using task queues in webrtc........
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/2289203002
Cr-Commit-Position: refs/heads/master@{#13983}
Log the DTLS handshake error code in OpenSSLStreamAdapter.
Forward the error code to WebRTCSession with the Signals.
This part is only for the WebRTC native code.
To make it work, need another CL for Chromium.
BUG=webrtc:5959
Review-Url: https://codereview.webrtc.org/2167363002
Cr-Commit-Position: refs/heads/master@{#13940}
The code that extracts certificate stats from an SSLCertificate and its
certificate chain is moved into SSLCertificate::GetStats. The stats
collector code loops through the resulting SSLCertificateStats and
creates the StatsReports for those stats.
This will allow the new stats collector to reuse GetStats in a future
CL.
BUG=chromium:627816, chromium:629436
Review-Url: https://codereview.webrtc.org/2259283002
Cr-Commit-Position: refs/heads/master@{#13917}
So that we don't have to use assert(). Includes one sample call site.
NOTRY=true
BUG=chromium:617124
Review-Url: https://codereview.webrtc.org/2262173002
Cr-Commit-Position: refs/heads/master@{#13862}
tests.
Note: The webrtc/base/test/ folder is new.
Currently not used, I intend to use this in another CL.
BUG=chromium:627816
NOPRESUBMIT=TRUE
NOTRY=TRUE
Review-Url: https://codereview.webrtc.org/2238073003
Cr-Commit-Position: refs/heads/master@{#13750}
The test worked by sleeping a certain time, then checking that the
difference between recv timestamps before and after the sleep was
within some margin of the requested sleep time.
However, this means that imprecision of SleepMs makes the test flaky.
This source of flakiness can be removed by comparing to the actual
time slept instead of the requested time.
Also making the margin larger, to further reduce the likelihood of
flakiness.
R=pthatcher@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2111043004 .
Cr-Commit-Position: refs/heads/master@{#13733}
magjed@ is the author of two out of three files in webrtc/base/java
(ThreadUtils.java and Size.java). They were originally created in
webrtc/api/android/java, but were moved to base/ in order to be
accessible from e.g. webrtc/modules/audio_device/android/java.
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/2233793002 .
Cr-Commit-Position: refs/heads/master@{#13707}
This prevents biased modulo arithmetic when selecting a character for a
random value from the provided table.
BUG=webrtc:5870
Review-Url: https://codereview.webrtc.org/2115793003
Cr-Commit-Position: refs/heads/master@{#13683}
RTC_CHECK in functions that return the random value.
BUG=webrtc:6072
Review-Url: https://codereview.webrtc.org/2119003002
Cr-Commit-Position: refs/heads/master@{#13682}
GCM cipher suites are optional (disabled by default) and can be enabled
through "PeerConnectionFactoryInterface::Options".
If compiled with Chromium (i.e. "ENABLE_EXTERNAL_AUTH" is defined), no
GCM ciphers can be used yet (see https://crbug.com/628400).
BUG=webrtc:5222, 628400
Review-Url: https://codereview.webrtc.org/1528843005
Cr-Commit-Position: refs/heads/master@{#13635}
This change makes WebRTC no longer stop sending video when we receive an
EWOULDBLOCK error from the operating system. This was previously
causing calls on a slow link (where the first hop is slow) to rapidly
oscillate between starting and stopping video.
We still do need to stop sending packets if there is no known good
connection we can use for that. We used to generate a synthetic
EWOULDBLOCK error in that case. This CL replaces it with a different
code (ENOTCONN); EWOULDBLOCK no longer stops the stream but ENOTCONN
does.
I've updated all the places where we seemed to be generating EWOULDBLOCK
for reasons other than some buffer been full; please give it a thorough
look in case I missed something.
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2192963002 .
Cr-Commit-Position: refs/heads/master@{#13566}
This CL fixes a crash that could happen when JSON event tracing is
shutting down. The cause of the crash was the fact that the logger
thread function was returning 'true', causing the platform thread to run
it repeatedly even though that wasn't the intention.
Usually the EventLogger::Stop() function would set the event requesting
the logging thread to clean up and close the file, and then immediately
call PlatformThread::Stop() which would stop the outer loop. The Log()
function would only run once and everything behaves as expected.
However, if a context switch happens between the shutdown_event_.Set()
and logging_thread_.Stop() calls in EventLogger::Stop(), the logger
thread function would close the file and exit the Log() method, while
PlatformThread will rerun it again. So the Log() function runs twice,
and the second time output_file_ is NULL which either causes the DCHECK
to fail (in debug builds) or the fprintf() to crash with SIGSEGV (in
release builds).
The fix simply changes the return value of the thread function to false
so it never runs twice.
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2168283002 .
Cr-Commit-Position: refs/heads/master@{#13510}
