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Issue 2380683005: Moved RtcEventLog files from call/ to logging/ (new top level dir) (Closed)
Patch Set: Updated GYP files Created 4 years, 2 months ago
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1 # Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 1 # Copyright (c) 2014 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("audio_coding.gni") 10 import("audio_coding.gni")
(...skipping 99 matching lines...)
110 cflags = [ 110 cflags = [
111 # TODO(kjellander): Bug 261: fix this warning. 111 # TODO(kjellander): Bug 261: fix this warning.
112 "/wd4373", # virtual function override. 112 "/wd4373", # virtual function override.
113 ] 113 ]
114 } 114 }
115 115
116 deps = audio_coding_deps + [ 116 deps = audio_coding_deps + [
117 ":audio_network_adaptor", 117 ":audio_network_adaptor",
118 ":neteq", 118 ":neteq",
119 ":rent_a_codec", 119 ":rent_a_codec",
120 "../../call:rtc_event_log", 120 "../../logging:rtc_event_log_api",
121 ] 121 ]
122 defines = audio_coding_defines 122 defines = audio_coding_defines
123 } 123 }
124 124
125 rtc_static_library("audio_decoder_interface") { 125 rtc_static_library("audio_decoder_interface") {
126 sources = [ 126 sources = [
127 "codecs/audio_decoder.cc", 127 "codecs/audio_decoder.cc",
128 "codecs/audio_decoder.h", 128 "codecs/audio_decoder.h",
129 "codecs/legacy_encoded_audio_frame.cc", 129 "codecs/legacy_encoded_audio_frame.cc",
130 "codecs/legacy_encoded_audio_frame.h", 130 "codecs/legacy_encoded_audio_frame.h",
(...skipping 876 matching lines...)
1007 "neteq/tools/rtc_event_log_source.h", 1007 "neteq/tools/rtc_event_log_source.h",
1008 ] 1008 ]
1009 1009
1010 if (is_clang) { 1010 if (is_clang) {
1011 # Suppress warnings from the Chromium Clang plugins 1011 # Suppress warnings from the Chromium Clang plugins
1012 # (bugs.webrtc.org/163). 1012 # (bugs.webrtc.org/163).
1013 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] 1013 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
1014 } 1014 }
1015 1015
1016 deps = [ 1016 deps = [
1017 "../../:rtc_event_log_parser", 1017 "../../logging:rtc_event_log_parser",
1018 ] 1018 ]
1019 public_deps = [ 1019 public_deps = [
1020 "../../:rtc_event_log_proto", 1020 "../../logging:rtc_event_log_proto",
1021 ] 1021 ]
1022 } 1022 }
1023 1023
1024 rtc_test("neteq_rtpplay") { 1024 rtc_test("neteq_rtpplay") {
1025 testonly = true 1025 testonly = true
1026 defines = [] 1026 defines = []
1027 deps = [] 1027 deps = []
1028 sources = [ 1028 sources = [
1029 "neteq/tools/neteq_rtpplay.cc", 1029 "neteq/tools/neteq_rtpplay.cc",
1030 ] 1030 ]
(...skipping 559 matching lines...)
1590 "//testing/gtest", 1590 "//testing/gtest",
1591 ] 1591 ]
1592 1592
1593 if (is_clang) { 1593 if (is_clang) {
1594 # Suppress warnings from Chrome's Clang plugins. 1594 # Suppress warnings from Chrome's Clang plugins.
1595 # See http://code.google.com/p/webrtc/issues/detail?id=163 for details. 1595 # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
1596 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] 1596 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
1597 } 1597 }
1598 } 1598 }
1599 } 1599 }
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