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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("//testing/test.gni") | 10 import("//testing/test.gni") |
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22 "include/voe_codec.h", | 22 "include/voe_codec.h", |
23 "include/voe_errors.h", | 23 "include/voe_errors.h", |
24 "include/voe_external_media.h", | 24 "include/voe_external_media.h", |
25 "include/voe_file.h", | 25 "include/voe_file.h", |
26 "include/voe_hardware.h", | 26 "include/voe_hardware.h", |
27 "include/voe_neteq_stats.h", | 27 "include/voe_neteq_stats.h", |
28 "include/voe_network.h", | 28 "include/voe_network.h", |
29 "include/voe_rtp_rtcp.h", | 29 "include/voe_rtp_rtcp.h", |
30 "include/voe_video_sync.h", | 30 "include/voe_video_sync.h", |
31 "include/voe_volume_control.h", | 31 "include/voe_volume_control.h", |
32 "level_indicator.cc", | |
33 "level_indicator.h", | |
34 "monitor_module.cc", | 32 "monitor_module.cc", |
35 "monitor_module.h", | 33 "monitor_module.h", |
36 "network_predictor.cc", | 34 "network_predictor.cc", |
37 "network_predictor.h", | 35 "network_predictor.h", |
38 "output_mixer.cc", | 36 "output_mixer.cc", |
39 "output_mixer.h", | 37 "output_mixer.h", |
40 "shared_data.cc", | 38 "shared_data.cc", |
41 "shared_data.h", | 39 "shared_data.h", |
42 "statistics.cc", | 40 "statistics.cc", |
43 "statistics.h", | 41 "statistics.h", |
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84 configs += [ "..:common_config" ] | 82 configs += [ "..:common_config" ] |
85 public_configs = [ "..:common_inherited_config" ] | 83 public_configs = [ "..:common_inherited_config" ] |
86 | 84 |
87 if (is_clang) { | 85 if (is_clang) { |
88 # Suppress warnings from Chrome's Clang plugins. | 86 # Suppress warnings from Chrome's Clang plugins. |
89 # See http://code.google.com/p/webrtc/issues/detail?id=163 for details. | 87 # See http://code.google.com/p/webrtc/issues/detail?id=163 for details. |
90 configs -= [ "//build/config/clang:find_bad_constructs" ] | 88 configs -= [ "//build/config/clang:find_bad_constructs" ] |
91 } | 89 } |
92 | 90 |
93 deps = [ | 91 deps = [ |
| 92 ":level_indicator", |
94 "..:rtc_event_log", | 93 "..:rtc_event_log", |
95 "..:webrtc_common", | 94 "..:webrtc_common", |
96 "../base:rtc_base_approved", | 95 "../base:rtc_base_approved", |
97 "../common_audio", | 96 "../common_audio", |
98 "../modules/audio_coding", | 97 "../modules/audio_coding", |
99 "../modules/audio_conference_mixer", | 98 "../modules/audio_conference_mixer", |
100 "../modules/audio_device", | 99 "../modules/audio_device", |
101 "../modules/audio_processing", | 100 "../modules/audio_processing", |
102 "../modules/bitrate_controller", | 101 "../modules/bitrate_controller", |
103 "../modules/media_file", | 102 "../modules/media_file", |
104 "../modules/pacing", | 103 "../modules/pacing", |
105 "../modules/rtp_rtcp", | 104 "../modules/rtp_rtcp", |
106 "../modules/utility", | 105 "../modules/utility", |
107 "../system_wrappers", | 106 "../system_wrappers", |
108 ] | 107 ] |
109 } | 108 } |
110 | 109 |
| 110 source_set("level_indicator") { |
| 111 sources = [ |
| 112 "level_indicator.cc", |
| 113 "level_indicator.h", |
| 114 ] |
| 115 |
| 116 configs += [ "..:common_config" ] |
| 117 public_configs = [ "..:common_inherited_config" ] |
| 118 |
| 119 deps = [ |
| 120 "..:webrtc_common", |
| 121 "../base:rtc_base_approved", |
| 122 "../common_audio", |
| 123 ] |
| 124 } |
| 125 |
111 if (rtc_include_tests) { | 126 if (rtc_include_tests) { |
112 test("voice_engine_unittests") { | 127 test("voice_engine_unittests") { |
113 deps = [ | 128 deps = [ |
114 ":voice_engine", | 129 ":voice_engine", |
115 "//testing/gmock", | 130 "//testing/gmock", |
116 "//testing/gtest", | 131 "//testing/gtest", |
117 "//webrtc/common_audio", | 132 "//webrtc/common_audio", |
118 "//webrtc/modules/audio_coding", | 133 "//webrtc/modules/audio_coding", |
119 "//webrtc/modules/audio_conference_mixer", | 134 "//webrtc/modules/audio_conference_mixer", |
120 "//webrtc/modules/audio_device", | 135 "//webrtc/modules/audio_device", |
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246 } | 261 } |
247 | 262 |
248 if (is_clang) { | 263 if (is_clang) { |
249 # Suppress warnings from Chrome's Clang plugins. | 264 # Suppress warnings from Chrome's Clang plugins. |
250 # See http://code.google.com/p/webrtc/issues/detail?id=163 for details. | 265 # See http://code.google.com/p/webrtc/issues/detail?id=163 for details. |
251 configs -= [ "//build/config/clang:find_bad_constructs" ] | 266 configs -= [ "//build/config/clang:find_bad_constructs" ] |
252 } | 267 } |
253 } | 268 } |
254 } | 269 } |
255 } | 270 } |
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