Remove a large number of targets that are no longer built, to reduce maintenance.
Only targets that have a GN version were removed.
BUG=webrtc:6323
NOTRY=True
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2340773003
Cr-Commit-Position: refs/heads/master@{#14231}
The old and new getStats are very different. This CL proposes rewriting
the new getStats from scratch with a bottom-up approach, starting with
the fundamental stats classes. This will allow cleaner and more
efficient code that is more aligned with the spec.
RTCStats and subclasses are the equivalent to RTCStats and RTCStats-
-derived dictionaries from the specs[1][2]. The dictionary members are
public member variables of type RTCStatsMember<T>, where T is one of the
supported types. All members derive from RTCStatsMemberInterface and
iteration of members is possible with RTCStats::Members().
The members are not stored in a map for performance and readability.
Type checking is supported with static class variables, kType.
Only the supported member types T are specialized and may be
instantiated, and sequences are supported with std::vector<...>. Type
checking is again supported with static class variables, kType.
RTCStatsReport is the equivalent from the spec[3], and maps RTCStats::id
to RTCStats-objects. RTCStatsReport is reference counted. It and its
contained stats may be destroyed on any thread. When the
RTCStatsCollector is added in a follow-up CL, it will return const
references to the RTCStatsReports. This means copies don't have to be
made for multiple stats observers or when jumping threads. In fact, no
copies of any stats will have to be made in surfacing stats to Blink.
[1] https://www.w3.org/TR/2016/WD-webrtc-20160531/#rtcstats-dictionary
[2] https://w3c.github.io/webrtc-stats/archives/20160526/webrtc-stats.html
[3] https://www.w3.org/TR/2016/WD-webrtc-20160531/#rtcstatsreport-object
This adds the new folder webrtc/stats/, with target rtc_stats and binary
rtc_stats_unittests. Public api headers are placed in webrtc/api/ and
.cc files are placed in webrtc/stats/.
BUG=chromium:627816
Review-Url: https://codereview.webrtc.org/2241093002
Cr-Commit-Position: refs/heads/master@{#13879}
Move the webrtc/test/test_support/metrics sources into
test_support[_unittests] targets.
This is essentially reverting https://webrtc-codereview.appspot.com/5789004
and moving these sources back to the right target.
Add missing foreman_cif.yuv resource needed for these tests.
For MIPS, a compile error was surfacing for logcat_trace_context.h when
flipping bot to GN, which was fixed.
BUG=webrtc:5949
NOTRY=True
Review-Url: https://codereview.webrtc.org/2267113002
Cr-Commit-Position: refs/heads/master@{#13860}
In order to switch Chromium to use WebRTC targets instead of
duplicated code listings in src/third_party/libjingle it must
be possible for Chromium to process webrtc/api/api.gyp. This is
currently not possible since it includes build/java.gypi, of which
the path is different in a Chromium checkout. It's not possible
to resolve this in another way since 'includes' processing takes
place early in the GYP cycle, before it's possible to use variables.
They're also processed ignoring conditional statements, resulting
in an error when api.gyp is processed.
BUG=webrtc:4256
TBR=perkj@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2080563002
Cr-Commit-Position: refs/heads/master@{#13208}
Move all its configuration to all.gyp instead, which is
not processed by Chromium builds (webrtc.gyp is).
BUG=webrtc:5949
NOTRY=True
Review-Url: https://codereview.webrtc.org/2060873002
Cr-Commit-Position: refs/heads/master@{#13124}
The legacy objc API is not included in the GYP generation if include_tests=0.
This causes problems downstream in some cases, so it's changed in this CL.
The libyuv dependency needs to be possible to disable using the build_libyuv
GYP variable.
NOTRY=True
Review URL: https://codereview.webrtc.org/1705733002
Cr-Commit-Position: refs/heads/master@{#11652}
In https://codereview.webrtc.org/1691463002 the legacy objc tests
were moved into a new GYP file, which was conditioned so it's not
included for platforms that cannot build it. This condition contained
an error which makes the GYP file being processed even if include_tests=0,
which causes errors in downstream code.
BUG=webrtc:5419
NOTRY=True
Review URL: https://codereview.webrtc.org/1701053005
Cr-Commit-Position: refs/heads/master@{#11650}
The JNI code for VoiceEngine is not maintained and VoiceEngine is being
refactored. This is not a supported Java interface, use AppRTCDemo as a
starting point instead.
Also renames webrtc/libjingle_examples.gyp webrtc/webrtc_examples.gyp to
replace the previous file (that only contained media_demo).
BUG=
R=henrika@webrtc.org, kjellander@webrtc.org
Review URL: https://codereview.webrtc.org/1439593002 .
Cr-Commit-Position: refs/heads/master@{#10599}
Significant changes:
- move the libjingle_examples.gyp file into webrtc directory.
- rename talk/examples/android to webrtc/examples/androidapp to avoid name conflicts.
- update paths in talk/libjingle_tests.gyp to point to webrtc directory for Objective-C test.
BUG=
R=pthatcher@webrtc.org, tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1235563006 .
Cr-Commit-Position: refs/heads/master@{#9681}
To make it possible to exclude the examples when running
GYP on all.gyp.
The webrtc_examples.gyp already has an OS=="android" condition
inside it, so there's no need to check that before including it.
BUG=webrtc:4242
Review URL: https://codereview.webrtc.org/1196623006
Cr-Commit-Position: refs/heads/master@{#9556}
Modifies the previous condition to additionally not use openmax_dl on
iOS. Remove the All target's direct dependency on it, as it is now
pulled in by the targets that need it.
Add gn support since an openmax_dl gn target is available.
BUG=chromium:415393, webrtc:3906
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7397 4adac7df-926f-26a2-2b94-8c16560cd09d
Things included in this CL:
- updated READMEs to provide an exact/reproable set of steps for getting the app
running.
- gyp changes to build the iOS AppRTCDemo sample app using gyp+ninja instead of
the hand-crafted Xcode project (which has never worked in its checked-in
form), including a gyp action to sign the sample app for deployment to an iOS
device (the app can also be used in the simulator)
- deleted the busted hand-crafted Xcode project for the sample app
- updated the sample app to match the PeerConnection API that ended up landing
(in a surprising twist of fate, the API landed quite a bit later than the
sample app and this is the first time the CR-time changes in the API are
reflected in the sample app)
- updated the sample app to reflect apprtc.appspot.com HTML/JS changes (equiv to
the AppRTCClient.java changes in http://s10/47299162)
- picked up the iossim DEPS to enable launching the sample app in the simulator
from the command-line.
- renamed some files to match capitalization of the classes they contain (Ice ->
ICE) per ObjC naming guidelines.
- ran the files involved in this CL through clang-format to deal with xcode
formatting craxy.
BUG=2106
RISK=P2
TESTED=unittest builds with ninja and passes on OS=mac; sample app builds with ninja and runs on simulator and device, though no audio flows from simulator/device (will fix in a follow-up CL)
R=andrew@webrtc.org, justincohen@google.com, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1874005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4466 4adac7df-926f-26a2-2b94-8c16560cd09d
