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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 | 10 |
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| 82 } | 82 } |
| 83 | 83 |
| 84 public_deps = [ | 84 public_deps = [ |
| 85 "../modules/audio_coding", | 85 "../modules/audio_coding", |
| 86 ] | 86 ] |
| 87 deps = [ | 87 deps = [ |
| 88 ":level_indicator", | 88 ":level_indicator", |
| 89 "..:webrtc_common", | 89 "..:webrtc_common", |
| 90 "../api:audio_mixer_api", | 90 "../api:audio_mixer_api", |
| 91 "../api:call_api", | 91 "../api:call_api", |
| 92 "../api:transport_api", |
| 92 "../base:rtc_base_approved", | 93 "../base:rtc_base_approved", |
| 93 "../common_audio", | 94 "../common_audio", |
| 94 "../logging:rtc_event_log_api", | 95 "../logging:rtc_event_log_api", |
| 95 "../modules/audio_coding:audio_decoder_factory_interface", | 96 "../modules/audio_coding:audio_decoder_factory_interface", |
| 96 "../modules/audio_coding:audio_format_conversion", | 97 "../modules/audio_coding:audio_format_conversion", |
| 97 "../modules/audio_coding:builtin_audio_decoder_factory", | 98 "../modules/audio_coding:builtin_audio_decoder_factory", |
| 98 "../modules/audio_coding:rent_a_codec", | 99 "../modules/audio_coding:rent_a_codec", |
| 99 "../modules/audio_conference_mixer", | 100 "../modules/audio_conference_mixer", |
| 100 "../modules/audio_device", | 101 "../modules/audio_device", |
| 101 "../modules/audio_processing", | 102 "../modules/audio_processing", |
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| 170 | 171 |
| 171 configs += [ ":channel_transport_warnings_config" ] | 172 configs += [ ":channel_transport_warnings_config" ] |
| 172 | 173 |
| 173 if (!build_with_chromium && is_clang) { | 174 if (!build_with_chromium && is_clang) { |
| 174 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 175 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| 175 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 176 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| 176 } | 177 } |
| 177 | 178 |
| 178 deps = [ | 179 deps = [ |
| 179 "..:webrtc_common", | 180 "..:webrtc_common", |
| 181 "../api:transport_api", |
| 180 "../system_wrappers", | 182 "../system_wrappers", |
| 181 "//testing/gtest", | 183 "//testing/gtest", |
| 182 ] | 184 ] |
| 183 } | 185 } |
| 184 | 186 |
| 185 rtc_test("voice_engine_unittests") { | 187 rtc_test("voice_engine_unittests") { |
| 186 deps = [ | 188 deps = [ |
| 187 ":channel_transport", | 189 ":channel_transport", |
| 188 ":voice_engine", | 190 ":voice_engine", |
| 189 "//testing/gmock", | 191 "//testing/gmock", |
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| 345 ] | 347 ] |
| 346 } | 348 } |
| 347 | 349 |
| 348 if (!build_with_chromium && is_clang) { | 350 if (!build_with_chromium && is_clang) { |
| 349 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163)
. | 351 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163)
. |
| 350 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 352 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| 351 } | 353 } |
| 352 } | 354 } |
| 353 } | 355 } |
| 354 } | 356 } |
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