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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("../webrtc.gni") | 9 import("../webrtc.gni") |
10 if (is_android) { | 10 if (is_android) { |
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32 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 32 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
33 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 33 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
34 } | 34 } |
35 | 35 |
36 deps = [ | 36 deps = [ |
37 "..:webrtc_common", | 37 "..:webrtc_common", |
38 "../api:audio_mixer_api", | 38 "../api:audio_mixer_api", |
39 "../api:call_api", | 39 "../api:call_api", |
40 "../api/audio_codecs:audio_codecs_api", | 40 "../api/audio_codecs:audio_codecs_api", |
41 "../api/audio_codecs:builtin_audio_encoder_factory", | 41 "../api/audio_codecs:builtin_audio_encoder_factory", |
42 "../base:rtc_base_approved", | |
43 "../base:rtc_task_queue", | |
44 "../call:call_interfaces", | 42 "../call:call_interfaces", |
45 "../call:rtp_interfaces", | 43 "../call:rtp_interfaces", |
46 "../common_audio", | 44 "../common_audio", |
47 "../modules/audio_coding:cng", | 45 "../modules/audio_coding:cng", |
48 "../modules/audio_device", | 46 "../modules/audio_device", |
49 "../modules/audio_processing", | 47 "../modules/audio_processing", |
50 "../modules/bitrate_controller:bitrate_controller", | 48 "../modules/bitrate_controller:bitrate_controller", |
51 "../modules/congestion_controller:congestion_controller", | 49 "../modules/congestion_controller:congestion_controller", |
52 "../modules/pacing:pacing", | 50 "../modules/pacing:pacing", |
53 "../modules/remote_bitrate_estimator:remote_bitrate_estimator", | 51 "../modules/remote_bitrate_estimator:remote_bitrate_estimator", |
54 "../modules/rtp_rtcp:rtp_rtcp", | 52 "../modules/rtp_rtcp:rtp_rtcp", |
| 53 "../rtc_base:rtc_base_approved", |
| 54 "../rtc_base:rtc_task_queue", |
55 "../system_wrappers", | 55 "../system_wrappers", |
56 "../voice_engine", | 56 "../voice_engine", |
57 ] | 57 ] |
58 } | 58 } |
59 if (rtc_include_tests) { | 59 if (rtc_include_tests) { |
60 rtc_source_set("audio_tests") { | 60 rtc_source_set("audio_tests") { |
61 testonly = true | 61 testonly = true |
62 | 62 |
63 # Skip restricting visibility on mobile platforms since the tests on those | 63 # Skip restricting visibility on mobile platforms since the tests on those |
64 # gets additional generated targets which would require many lines here to | 64 # gets additional generated targets which would require many lines here to |
65 # cover (which would be confusing to read and hard to maintain). | 65 # cover (which would be confusing to read and hard to maintain). |
66 if (!is_android && !is_ios) { | 66 if (!is_android && !is_ios) { |
67 visibility = [ "..:video_engine_tests" ] | 67 visibility = [ "..:video_engine_tests" ] |
68 } | 68 } |
69 | 69 |
70 # TODO(kjellander): Remove (bugs.webrtc.org/6828) | 70 # TODO(kjellander): Remove (bugs.webrtc.org/6828) |
71 # This needs remote_bitrate_estimator to be moved to webrtc/api first. | 71 # This needs remote_bitrate_estimator to be moved to webrtc/api first. |
72 check_includes = false | 72 check_includes = false |
73 | 73 |
74 sources = [ | 74 sources = [ |
75 "audio_receive_stream_unittest.cc", | 75 "audio_receive_stream_unittest.cc", |
76 "audio_send_stream_unittest.cc", | 76 "audio_send_stream_unittest.cc", |
77 "audio_state_unittest.cc", | 77 "audio_state_unittest.cc", |
78 "time_interval_unittest.cc", | 78 "time_interval_unittest.cc", |
79 ] | 79 ] |
80 deps = [ | 80 deps = [ |
81 ":audio", | 81 ":audio", |
82 "../api:mock_audio_mixer", | 82 "../api:mock_audio_mixer", |
83 "../base:rtc_base_approved", | |
84 "../base:rtc_task_queue", | |
85 "../call:rtp_receiver", | 83 "../call:rtp_receiver", |
86 "../modules/audio_device:mock_audio_device", | 84 "../modules/audio_device:mock_audio_device", |
87 "../modules/audio_mixer:audio_mixer_impl", | 85 "../modules/audio_mixer:audio_mixer_impl", |
88 "../modules/congestion_controller:congestion_controller", | 86 "../modules/congestion_controller:congestion_controller", |
89 "../modules/congestion_controller:mock_congestion_controller", | 87 "../modules/congestion_controller:mock_congestion_controller", |
90 "../modules/pacing:pacing", | 88 "../modules/pacing:pacing", |
| 89 "../rtc_base:rtc_base_approved", |
| 90 "../rtc_base:rtc_task_queue", |
91 "../test:test_common", | 91 "../test:test_common", |
92 "../test:test_support", | 92 "../test:test_support", |
93 "utility:utility_tests", | 93 "utility:utility_tests", |
94 "//testing/gmock", | 94 "//testing/gmock", |
95 "//testing/gtest", | 95 "//testing/gtest", |
96 ] | 96 ] |
97 | 97 |
98 if (!build_with_chromium && is_clang) { | 98 if (!build_with_chromium && is_clang) { |
99 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 99 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
100 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 100 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
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144 # gets additional generated targets which would require many lines here to | 144 # gets additional generated targets which would require many lines here to |
145 # cover (which would be confusing to read and hard to maintain). | 145 # cover (which would be confusing to read and hard to maintain). |
146 if (!is_android && !is_ios) { | 146 if (!is_android && !is_ios) { |
147 visibility = [ "//webrtc:webrtc_perf_tests" ] | 147 visibility = [ "//webrtc:webrtc_perf_tests" ] |
148 } | 148 } |
149 sources = [ | 149 sources = [ |
150 "test/audio_bwe_integration_test.cc", | 150 "test/audio_bwe_integration_test.cc", |
151 "test/audio_bwe_integration_test.h", | 151 "test/audio_bwe_integration_test.h", |
152 ] | 152 ] |
153 deps = [ | 153 deps = [ |
154 "../base:rtc_base_approved", | |
155 "../common_audio", | 154 "../common_audio", |
| 155 "../rtc_base:rtc_base_approved", |
156 "../system_wrappers", | 156 "../system_wrappers", |
157 "../test:fake_audio_device", | 157 "../test:fake_audio_device", |
158 "../test:field_trial", | 158 "../test:field_trial", |
159 "../test:test_common", | 159 "../test:test_common", |
160 "../test:test_main", | 160 "../test:test_main", |
161 "//testing/gmock", | 161 "//testing/gmock", |
162 "//testing/gtest", | 162 "//testing/gtest", |
163 "//third_party/gflags", | 163 "//third_party/gflags", |
164 ] | 164 ] |
165 | 165 |
166 data = [ | 166 data = [ |
167 "//resources/voice_engine/audio_dtx16.wav", | 167 "//resources/voice_engine/audio_dtx16.wav", |
168 ] | 168 ] |
169 | 169 |
170 if (!build_with_chromium && is_clang) { | 170 if (!build_with_chromium && is_clang) { |
171 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 171 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
172 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 172 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
173 } | 173 } |
174 } | 174 } |
175 } | 175 } |
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