The reason for this is that http://crrev.com/245412
introduces a dependency of Chrome's src/build/gyp_chromium
to src/tools/find_depot_tools.py, which we don't have
synced in WebRTC (src/tools is very big).
Offline discussions shows that we cannot rely on syncing
individual subdirectories from Chrome in the future, but
maintaining our own gyp_webrtc file will at least buy us
some time for now, so we can roll past that chromium_revision
in WebRTC DEPS.
Overview of differences between gyp_webrtc and gyp_chromium
(and how we previously used gyp_chromium):
* No .gyp file needs to be passed (defaults to all.gyp)
* CHROMIUM_GYP_FILE is ignored (i.e. cannot be used to
specify an alternate .gyp file to process)
* Ninja is used by default on all platforms unless GYP_GENERATORS
is set.
* Gyp syntax check is always on
* Gyp circular dependency check is always on
* No support for automatic toolchain detection on Windows.
* --depth argument is no longer needed since calculated by
the script.
* Support for a webrtc.gyp_env file sitting next to the
.gclient file in the top dir of checkout, which can be
used to override Gyp variables similar to chromium.gyp_env.
* SKIP_WEBRTC_GYP_ENV can be set to skip reading webrtc.gyp_env.
BUG=2863
TEST=Ran and verified behavior on Linux with:
gclient runhooks
webrtc/build/gyp_webrtc
webrtc/build/gyp_webrtc -Dextra_gyp_flag=0
. build/android/envsetup.sh && gclient runhooks
SKIP_WEBRTC_GYP_ENV=1 webrtc/build/gyp_webrtc
GYP_GENERATORS=make webrtc/build/gyp_webrtc
The patch also passes runhooks and compile step on all trybots.
R=andrew@webrtc.org, fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7759004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5467 4adac7df-926f-26a2-2b94-8c16560cd09d
This will allow an embedder to use it directly.
Adding inertia/hangover time between updates of the reported detection status to the algorithm, controlled by a parameter. That is usually desired and this way a consumer of
the class don't have to implement that. (VoiceEngine will let it be 1, which results in the same behavior as before, and keep controlling the hangover itself.)
R=andrew@webrtc.org, niklas.enbom@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5462 4adac7df-926f-26a2-2b94-8c16560cd09d
It includes unit test fixes to properly initialize SSL if DTLS or SSL random number generator is used in the tests.
The private key and certificate constant strings used in some tests are updated to be compatible with NSS.
A few potentially overflow type conversions caused compiling warning on Windows and they are fixed by importing and using Chromium's checked_cast, which aborts the program if overflow occurs.
It also fixes a leak in nssstreamadapter.cc by releasing the PRFileDesc* in StreamClose.
BUG=2253
R=fischman@webrtc.org, juberti@google.com, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4679005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5459 4adac7df-926f-26a2-2b94-8c16560cd09d
In webrtc::vcm::VideoReceiver::ResetDecoder(), the lock order is:
1. take _receiveCritSect,
2. take process_crit_sect_
This conflicts with the follow code path:
1. webrtc::vcm::VideoReceiver::Process(), take process_crit_sect_
call -> webrtc::vcm::VideoReceiver::NackList(),
2. with nackStats=kNackKeyFrameRequest, take _receiveCritSect
BUG=2861
TEST=trybots
R=sprang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5456 4adac7df-926f-26a2-2b94-8c16560cd09d
Add another test to NetEqPerformanceTest with no packet losses or
clock drift. The purpose of this test would be to focus on the
"clean" code path, i.e., the path taken when there are no network
problems. The reason is that this code path is presumably much
lighter in complexity, and regressions could easily drown in the
heavier code involved when combating losses and drift.
BUG=2859
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7689005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5452 4adac7df-926f-26a2-2b94-8c16560cd09d
As the new bots building the WebRTC native tests for Android as APKs
and executing them on a device has now proven to be reasonably stable,
it is time to enable them by default for tryjobs.
TEST=several green builds sent from a WebRTC checkout.
BUG=chromium:312827
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5440 4adac7df-926f-26a2-2b94-8c16560cd09d
There are uninitializion problem with normal_asyn_test.cc. This is fairly easy to solve and therefore is included in this CL.
The following is a memo on the selection of the version to roll. It may be a reference for similar missions.
How was this version picked?
1. The whole purpose of this work is to update to Clang to be able to compile Opus 1.1. In Chromium, Clang got updated to 198389 at r244540.
2. From r245412, gyp_chromium requires "tools\find_depot_tools.py". However, WebRTC does not sync up the root of folder "tools". An issue has been created to Chromium on this.
... So the version must be a good version between r244540 and r245411 (inclusive)
BUG=
TEST=passes all trybots
R=kjellander@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7569005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5436 4adac7df-926f-26a2-2b94-8c16560cd09d
This allows a listener to receive new statistics (byte/packet counts, etc) as it
is generated - avoiding the need to poll. This also makes handling stats from
multiple RTP streams more tractable. The change is primarily targeted at the new
video engine API.
TEST=Unit test in ReceiveStatisticsTest.
Integration tests to follow as call tests when fully wired up.
BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6259004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5416 4adac7df-926f-26a2-2b94-8c16560cd09d
