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Issue 2800633004: Resolve dependency between rtc_event_log_api and remote_bitrate_estimator (Closed)
Patch Set: Rebased Created 3 years, 8 months ago
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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")
11 if (is_android) { 11 if (is_android) {
12 import("//build/config/android/config.gni") 12 import("//build/config/android/config.gni")
13 import("//build/config/android/rules.gni") 13 import("//build/config/android/rules.gni")
14 } 14 }
15 15
16 group("logging") { 16 group("logging") {
17 public_deps = [ 17 public_deps = [
18 ":rtc_event_log_impl", 18 ":rtc_event_log_impl",
19 ] 19 ]
20 if (rtc_enable_protobuf) { 20 if (rtc_enable_protobuf) {
21 public_deps += [ ":rtc_event_log_parser" ] 21 public_deps += [ ":rtc_event_log_parser" ]
22 } 22 }
23 } 23 }
24 24
25 rtc_source_set("rtc_event_log_api") { 25 rtc_source_set("rtc_event_log_api") {
26 # TODO(kjellander): Remove (bugs.webrtc.org/7257)
27 # Enabling GN check triggers cyclic dependency error:
28 # //webrtc/logging:rtc_event_log_api ->
29 # //webrtc/modules/audio_coding:audio_network_adaptor ->
30 # //webrtc/logging:rtc_event_log_api
31 check_includes = false
32 sources = [ 26 sources = [
33 "rtc_event_log/rtc_event_log.h", 27 "rtc_event_log/rtc_event_log.h",
34 ] 28 ]
35 deps = [ 29 deps = [
36 "../base:rtc_base_approved", 30 "../base:rtc_base_approved",
37 "../call:call_interfaces", 31 "../call:call_interfaces",
38 ] 32 ]
39 } 33 }
40 34
41 rtc_static_library("rtc_event_log_impl") { 35 rtc_static_library("rtc_event_log_impl") {
42 sources = [ 36 sources = [
43 "rtc_event_log/ringbuffer.h", 37 "rtc_event_log/ringbuffer.h",
44 "rtc_event_log/rtc_event_log.cc", 38 "rtc_event_log/rtc_event_log.cc",
45 "rtc_event_log/rtc_event_log_helper_thread.cc", 39 "rtc_event_log/rtc_event_log_helper_thread.cc",
46 "rtc_event_log/rtc_event_log_helper_thread.h", 40 "rtc_event_log/rtc_event_log_helper_thread.h",
47 ] 41 ]
48 42
49 defines = [] 43 defines = []
50 44
51 deps = [ 45 deps = [
52 ":rtc_event_log_api", 46 ":rtc_event_log_api",
53 "..:webrtc_common", 47 "..:webrtc_common",
54 "../base:protobuf_utils", 48 "../base:protobuf_utils",
55 "../base:rtc_base_approved", 49 "../base:rtc_base_approved",
56 "../call:call_interfaces", 50 "../call:call_interfaces",
57 "../modules/audio_coding:audio_network_adaptor", 51 "../modules/audio_coding:audio_network_adaptor",
52 "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
58 "../modules/rtp_rtcp", 53 "../modules/rtp_rtcp",
59 "../system_wrappers", 54 "../system_wrappers",
60 ] 55 ]
61 56
62 if (rtc_enable_protobuf) { 57 if (rtc_enable_protobuf) {
63 defines += [ "ENABLE_RTC_EVENT_LOG" ] 58 defines += [ "ENABLE_RTC_EVENT_LOG" ]
64 deps += [ ":rtc_event_log_proto" ] 59 deps += [ ":rtc_event_log_proto" ]
65 } 60 }
66 if (!build_with_chromium && is_clang) { 61 if (!build_with_chromium && is_clang) {
67 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). 62 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
(...skipping 14 matching lines...) Expand all
82 "rtc_event_log/rtc_event_log_parser.cc", 77 "rtc_event_log/rtc_event_log_parser.cc",
83 "rtc_event_log/rtc_event_log_parser.h", 78 "rtc_event_log/rtc_event_log_parser.h",
84 ] 79 ]
85 80
86 public_deps = [ 81 public_deps = [
87 ":rtc_event_log_api", 82 ":rtc_event_log_api",
88 ":rtc_event_log_proto", 83 ":rtc_event_log_proto",
89 "..:webrtc_common", 84 "..:webrtc_common",
90 "../call:call_interfaces", 85 "../call:call_interfaces",
91 "../modules/audio_coding:audio_network_adaptor", 86 "../modules/audio_coding:audio_network_adaptor",
87 "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
92 "../modules/rtp_rtcp:rtp_rtcp", 88 "../modules/rtp_rtcp:rtp_rtcp",
93 "../system_wrappers", 89 "../system_wrappers",
94 ] 90 ]
95 91
96 if (!build_with_chromium && is_clang) { 92 if (!build_with_chromium && is_clang) {
97 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). 93 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
98 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] 94 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
99 } 95 }
100 deps = [ 96 deps = [
101 "../base:protobuf_utils", 97 "../base:protobuf_utils",
102 "../base:rtc_base_approved", 98 "../base:rtc_base_approved",
103 ] 99 ]
104 } 100 }
105 101
106 if (rtc_include_tests) { 102 if (rtc_include_tests) {
107 rtc_source_set("rtc_event_log_tests") { 103 rtc_source_set("rtc_event_log_tests") {
108 testonly = true 104 testonly = true
109 sources = [ 105 sources = [
110 "rtc_event_log/ringbuffer_unittest.cc", 106 "rtc_event_log/ringbuffer_unittest.cc",
111 "rtc_event_log/rtc_event_log_unittest.cc", 107 "rtc_event_log/rtc_event_log_unittest.cc",
112 "rtc_event_log/rtc_event_log_unittest_helper.cc", 108 "rtc_event_log/rtc_event_log_unittest_helper.cc",
113 ] 109 ]
114 deps = [ 110 deps = [
115 ":rtc_event_log_impl", 111 ":rtc_event_log_impl",
116 ":rtc_event_log_parser", 112 ":rtc_event_log_parser",
117 "../base:rtc_base_approved", 113 "../base:rtc_base_approved",
118 "../base:rtc_base_tests_utils", 114 "../base:rtc_base_tests_utils",
119 "../call", 115 "../call",
120 "../modules/audio_coding:audio_network_adaptor", 116 "../modules/audio_coding:audio_network_adaptor",
117 "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
121 "../modules/rtp_rtcp", 118 "../modules/rtp_rtcp",
122 "../system_wrappers:metrics_default", 119 "../system_wrappers:metrics_default",
123 "../test:test_support", 120 "../test:test_support",
124 "//testing/gmock", 121 "//testing/gmock",
125 "//testing/gtest", 122 "//testing/gtest",
126 ] 123 ]
127 if (!build_with_chromium && is_clang) { 124 if (!build_with_chromium && is_clang) {
128 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163) . 125 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163) .
129 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] 126 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
130 } 127 }
(...skipping 39 matching lines...) Expand 10 before | Expand all | Expand 10 after
170 "../modules/rtp_rtcp:rtp_rtcp", 167 "../modules/rtp_rtcp:rtp_rtcp",
171 "//third_party/gflags", 168 "//third_party/gflags",
172 ] 169 ]
173 if (!build_with_chromium && is_clang) { 170 if (!build_with_chromium && is_clang) {
174 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163) . 171 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163) .
175 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] 172 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
176 } 173 }
177 } 174 }
178 } 175 }
179 } 176 }
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