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1 # Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. | 1 # Copyright (c) 2016 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("../webrtc.gni") | 9 import("../webrtc.gni") |
10 import("//third_party/protobuf/proto_library.gni") | 10 import("//third_party/protobuf/proto_library.gni") |
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44 "rtc_event_log/rtc_event_log.cc", | 44 "rtc_event_log/rtc_event_log.cc", |
45 "rtc_event_log/rtc_event_log_helper_thread.cc", | 45 "rtc_event_log/rtc_event_log_helper_thread.cc", |
46 "rtc_event_log/rtc_event_log_helper_thread.h", | 46 "rtc_event_log/rtc_event_log_helper_thread.h", |
47 ] | 47 ] |
48 | 48 |
49 defines = [] | 49 defines = [] |
50 | 50 |
51 deps = [ | 51 deps = [ |
52 ":rtc_event_log_api", | 52 ":rtc_event_log_api", |
53 "..:webrtc_common", | 53 "..:webrtc_common", |
| 54 "../base:protobuf_utils", |
54 "../base:rtc_base_approved", | 55 "../base:rtc_base_approved", |
55 "../call:call_interfaces", | 56 "../call:call_interfaces", |
56 "../modules/audio_coding:audio_network_adaptor", | 57 "../modules/audio_coding:audio_network_adaptor", |
57 "../modules/rtp_rtcp", | 58 "../modules/rtp_rtcp", |
58 "../system_wrappers", | 59 "../system_wrappers", |
59 ] | 60 ] |
60 | 61 |
61 if (rtc_enable_protobuf) { | 62 if (rtc_enable_protobuf) { |
62 defines += [ "ENABLE_RTC_EVENT_LOG" ] | 63 defines += [ "ENABLE_RTC_EVENT_LOG" ] |
63 deps += [ ":rtc_event_log_proto" ] | 64 deps += [ ":rtc_event_log_proto" ] |
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90 "../modules/audio_coding:audio_network_adaptor", | 91 "../modules/audio_coding:audio_network_adaptor", |
91 "../modules/rtp_rtcp:rtp_rtcp", | 92 "../modules/rtp_rtcp:rtp_rtcp", |
92 "../system_wrappers", | 93 "../system_wrappers", |
93 ] | 94 ] |
94 | 95 |
95 if (!build_with_chromium && is_clang) { | 96 if (!build_with_chromium && is_clang) { |
96 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 97 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
97 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 98 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
98 } | 99 } |
99 deps = [ | 100 deps = [ |
| 101 "../base:protobuf_utils", |
100 "../base:rtc_base_approved", | 102 "../base:rtc_base_approved", |
101 ] | 103 ] |
102 } | 104 } |
103 | 105 |
104 if (rtc_include_tests) { | 106 if (rtc_include_tests) { |
105 rtc_source_set("rtc_event_log_tests") { | 107 rtc_source_set("rtc_event_log_tests") { |
106 testonly = true | 108 testonly = true |
107 sources = [ | 109 sources = [ |
108 "rtc_event_log/ringbuffer_unittest.cc", | 110 "rtc_event_log/ringbuffer_unittest.cc", |
109 "rtc_event_log/rtc_event_log_unittest.cc", | 111 "rtc_event_log/rtc_event_log_unittest.cc", |
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168 "../modules/rtp_rtcp:rtp_rtcp", | 170 "../modules/rtp_rtcp:rtp_rtcp", |
169 "//third_party/gflags", | 171 "//third_party/gflags", |
170 ] | 172 ] |
171 if (!build_with_chromium && is_clang) { | 173 if (!build_with_chromium && is_clang) { |
172 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163)
. | 174 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163)
. |
173 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 175 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
174 } | 176 } |
175 } | 177 } |
176 } | 178 } |
177 } | 179 } |
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