webrtc_m130/all.gyp
hbos 615d3013de RTCStats and RTCStatsReport added (webrtc/stats).
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}
2016-08-24 08:33:19 +00:00

82 lines
2.4 KiB
Python

# Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'variables': {
'include_examples%': 1,
'include_tests%': 1,
},
'includes': [
'webrtc/build/common.gypi',
],
'targets': [
{
'target_name': 'All',
'type': 'none',
'dependencies': [
'webrtc/api/api.gyp:*',
'webrtc/base/base.gyp:*',
'webrtc/common.gyp:*',
'webrtc/common_audio/common_audio.gyp:*',
'webrtc/common_video/common_video.gyp:*',
'webrtc/media/media.gyp:*',
'webrtc/modules/modules.gyp:*',
'webrtc/p2p/p2p.gyp:*',
'webrtc/pc/pc.gyp:*',
'webrtc/stats/stats.gyp:*',
'webrtc/system_wrappers/system_wrappers.gyp:*',
'webrtc/tools/tools.gyp:*',
'webrtc/voice_engine/voice_engine.gyp:*',
'webrtc/webrtc.gyp:*',
'<(webrtc_vp8_dir)/vp8.gyp:*',
'<(webrtc_vp9_dir)/vp9.gyp:*',
],
'conditions': [
['OS=="android" and build_with_chromium==0', {
'dependencies': [
'webrtc/api/api_java.gyp:*',
],
}],
['include_tests==1', {
'includes': [
'webrtc/webrtc_tests.gypi',
],
'dependencies': [
'webrtc/api/api_tests.gyp:*',
'webrtc/common_video/common_video_unittests.gyp:*',
'webrtc/system_wrappers/system_wrappers_tests.gyp:*',
'webrtc/test/test.gyp:*',
],
}],
['include_examples==1', {
'dependencies': [
'webrtc/webrtc_examples.gyp:*',
],
}],
['(OS=="ios" or (OS=="mac" and mac_deployment_target=="10.7"))', {
'dependencies': [
'webrtc/sdk/sdk.gyp:*',
],
}],
['OS=="ios" or (OS=="mac" and target_arch!="ia32")', {
'dependencies': [
'talk/app/webrtc/legacy_objc_api.gyp:*',
],
'conditions': [
['include_tests==1', {
'dependencies': [
'talk/app/webrtc/legacy_objc_api_tests.gyp:*',
],
}],
],
}],
],
},
],
}