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Issue 2242043002: RTCStatsCollector and RTCPeerConnectionStats added (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Changed ID of RTCPeerConnectionStats to "RTCPeerConnection" (no need to rerun bots yet) Created 4 years, 3 months ago
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1 # Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 1 # Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
2 # 2 #
3 # Use of this source code is governed by a BSD-style license 3 # Use of this source code is governed by a BSD-style license
4 # that can be found in the LICENSE file in the root of the source 4 # that can be found in the LICENSE file in the root of the source
5 # tree. An additional intellectual property rights grant can be found 5 # tree. An additional intellectual property rights grant can be found
6 # in the file PATENTS. All contributing project authors may 6 # in the file PATENTS. All contributing project authors may
7 # be found in the AUTHORS file in the root of the source tree. 7 # be found in the AUTHORS file in the root of the source tree.
8 8
9 import("../build/webrtc.gni") 9 import("../build/webrtc.gni")
10 import("//testing/test.gni") 10 import("//testing/test.gni")
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63 "peerconnection.h", 63 "peerconnection.h",
64 "peerconnectionfactory.cc", 64 "peerconnectionfactory.cc",
65 "peerconnectionfactory.h", 65 "peerconnectionfactory.h",
66 "peerconnectionfactoryproxy.h", 66 "peerconnectionfactoryproxy.h",
67 "peerconnectioninterface.h", 67 "peerconnectioninterface.h",
68 "peerconnectionproxy.h", 68 "peerconnectionproxy.h",
69 "proxy.h", 69 "proxy.h",
70 "remoteaudiosource.cc", 70 "remoteaudiosource.cc",
71 "remoteaudiosource.h", 71 "remoteaudiosource.h",
72 "rtcstats.h", 72 "rtcstats.h",
73 "rtcstats_objects.h",
73 "rtcstatsreport.h", 74 "rtcstatsreport.h",
74 "rtpparameters.h", 75 "rtpparameters.h",
75 "rtpreceiver.cc", 76 "rtpreceiver.cc",
76 "rtpreceiver.h", 77 "rtpreceiver.h",
77 "rtpreceiverinterface.h", 78 "rtpreceiverinterface.h",
78 "rtpsender.cc", 79 "rtpsender.cc",
79 "rtpsender.h", 80 "rtpsender.h",
80 "rtpsenderinterface.h", 81 "rtpsenderinterface.h",
81 "sctputils.cc", 82 "sctputils.cc",
82 "sctputils.h", 83 "sctputils.h",
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329 "rtpsenderreceiver_unittest.cc", 330 "rtpsenderreceiver_unittest.cc",
330 "statscollector_unittest.cc", 331 "statscollector_unittest.cc",
331 "test/fakeaudiocapturemodule.cc", 332 "test/fakeaudiocapturemodule.cc",
332 "test/fakeaudiocapturemodule.h", 333 "test/fakeaudiocapturemodule.h",
333 "test/fakeaudiocapturemodule_unittest.cc", 334 "test/fakeaudiocapturemodule_unittest.cc",
334 "test/fakeconstraints.h", 335 "test/fakeconstraints.h",
335 "test/fakedatachannelprovider.h", 336 "test/fakedatachannelprovider.h",
336 "test/fakeperiodicvideocapturer.h", 337 "test/fakeperiodicvideocapturer.h",
337 "test/fakertccertificategenerator.h", 338 "test/fakertccertificategenerator.h",
338 "test/fakevideotrackrenderer.h", 339 "test/fakevideotrackrenderer.h",
340 "test/mock_datachannel.h",
339 "test/mock_peerconnection.h", 341 "test/mock_peerconnection.h",
340 "test/mock_webrtcsession.h", 342 "test/mock_webrtcsession.h",
341 "test/mockpeerconnectionobservers.h", 343 "test/mockpeerconnectionobservers.h",
342 "test/peerconnectiontestwrapper.cc", 344 "test/peerconnectiontestwrapper.cc",
343 "test/peerconnectiontestwrapper.h", 345 "test/peerconnectiontestwrapper.h",
344 "test/testsdpstrings.h", 346 "test/testsdpstrings.h",
345 "videocapturertracksource_unittest.cc", 347 "videocapturertracksource_unittest.cc",
346 "videotrack_unittest.cc", 348 "videotrack_unittest.cc",
347 "webrtcsdp_unittest.cc", 349 "webrtcsdp_unittest.cc",
348 "webrtcsession_unittest.cc", 350 "webrtcsession_unittest.cc",
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439 441
440 shared_libraries = [ ":libjingle_peerconnection_so" ] 442 shared_libraries = [ ":libjingle_peerconnection_so" ]
441 } 443 }
442 444
443 android_resources("libjingle_peerconnection_android_unittest_resources") { 445 android_resources("libjingle_peerconnection_android_unittest_resources") {
444 resource_dirs = [ "androidtests/res" ] 446 resource_dirs = [ "androidtests/res" ]
445 custom_package = "org.webrtc" 447 custom_package = "org.webrtc"
446 } 448 }
447 } 449 }
448 } 450 }
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