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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("../../webrtc.gni") | 9 import("../../webrtc.gni") |
| 10 import("audio_coding.gni") | 10 import("audio_coding.gni") |
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| 75 "../../api/audio_codecs:audio_codecs_api", | 75 "../../api/audio_codecs:audio_codecs_api", |
| 76 "../..:webrtc_common", | 76 "../..:webrtc_common", |
| 77 "../../base:rtc_base_approved", | 77 "../../base:rtc_base_approved", |
| 78 "../../system_wrappers", | 78 "../../system_wrappers", |
| 79 ":audio_coding_module_typedefs", | 79 ":audio_coding_module_typedefs", |
| 80 ":audio_encoder_interface", | 80 ":audio_encoder_interface", |
| 81 ":isac_common", | 81 ":isac_common", |
| 82 ":isac_fix_c", | 82 ":isac_fix_c", |
| 83 ":neteq_decoder_enum", | 83 ":neteq_decoder_enum", |
| 84 ] + audio_codec_deps | 84 ] + audio_codec_deps |
| 85 | |
| 86 if (rtc_enable_protobuf) { | |
| 87 deps += [ "../../base:protobuf_utils" ] | |
| 88 } | |
|
kwiberg-webrtc
2017/03/22 10:03:23
Why would you ever want a conditional dep on proto
mbonadei
2017/03/23 16:13:58
Done and thanks for spotting this inconsistency. M
| |
| 89 | |
| 85 defines = audio_codec_defines | 90 defines = audio_codec_defines |
| 86 } | 91 } |
| 87 | 92 |
| 88 config("audio_coding_config") { | 93 config("audio_coding_config") { |
| 89 include_dirs = [ | 94 include_dirs = [ |
| 90 "include", | 95 "include", |
| 91 "../include", | 96 "../include", |
| 92 ] | 97 ] |
| 93 } | 98 } |
| 94 | 99 |
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| 830 "../../api/audio_codecs:audio_codecs_api", | 835 "../../api/audio_codecs:audio_codecs_api", |
| 831 "../../base:rtc_base_approved", | 836 "../../base:rtc_base_approved", |
| 832 "../../base:rtc_numerics", | 837 "../../base:rtc_numerics", |
| 833 "../../common_audio", | 838 "../../common_audio", |
| 834 "../../system_wrappers", | 839 "../../system_wrappers", |
| 835 ] | 840 ] |
| 836 public_deps = [ | 841 public_deps = [ |
| 837 ":webrtc_opus_c", | 842 ":webrtc_opus_c", |
| 838 ] | 843 ] |
| 839 | 844 |
| 845 if (rtc_enable_protobuf) { | |
| 846 deps += [ "../../base:protobuf_utils" ] | |
| 847 } | |
| 848 | |
| 840 defines = audio_codec_defines | 849 defines = audio_codec_defines |
| 841 if (rtc_opus_variable_complexity) { | 850 if (rtc_opus_variable_complexity) { |
| 842 defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=1" ] | 851 defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=1" ] |
| 843 } else { | 852 } else { |
| 844 defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=0" ] | 853 defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=0" ] |
| 845 } | 854 } |
| 846 | 855 |
| 847 if (rtc_build_opus) { | 856 if (rtc_build_opus) { |
| 848 public_deps += [ rtc_opus_dir ] | 857 public_deps += [ rtc_opus_dir ] |
| 849 } else if (build_with_mozilla) { | 858 } else if (build_with_mozilla) { |
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| 921 "../../base:rtc_base_approved", | 930 "../../base:rtc_base_approved", |
| 922 "../../common_audio", | 931 "../../common_audio", |
| 923 "../../logging:rtc_event_log_api", | 932 "../../logging:rtc_event_log_api", |
| 924 "../../system_wrappers", | 933 "../../system_wrappers", |
| 925 ] | 934 ] |
| 926 | 935 |
| 927 if (rtc_enable_protobuf) { | 936 if (rtc_enable_protobuf) { |
| 928 deps += [ | 937 deps += [ |
| 929 ":ana_config_proto", | 938 ":ana_config_proto", |
| 930 ":ana_debug_dump_proto", | 939 ":ana_debug_dump_proto", |
| 940 "../../base:protobuf_utils", | |
| 931 ] | 941 ] |
| 932 defines = [ "WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP" ] | 942 defines = [ "WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP" ] |
| 933 } | 943 } |
| 934 | 944 |
| 935 if (!build_with_chromium && is_clang) { | 945 if (!build_with_chromium && is_clang) { |
| 936 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 946 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| 937 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 947 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| 938 } | 948 } |
| 939 } | 949 } |
| 940 | 950 |
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| 1183 ] | 1193 ] |
| 1184 deps = [ | 1194 deps = [ |
| 1185 ":neteq_test_support", | 1195 ":neteq_test_support", |
| 1186 ":neteq_unittest_tools", | 1196 ":neteq_unittest_tools", |
| 1187 ":webrtc_opus", | 1197 ":webrtc_opus", |
| 1188 "../..:webrtc_common", | 1198 "../..:webrtc_common", |
| 1189 "../../base:rtc_base_approved", | 1199 "../../base:rtc_base_approved", |
| 1190 "../../system_wrappers:system_wrappers", | 1200 "../../system_wrappers:system_wrappers", |
| 1191 "../../test:test_support", | 1201 "../../test:test_support", |
| 1192 ] | 1202 ] |
| 1203 | |
| 1204 if (rtc_enable_protobuf) { | |
| 1205 deps += [ "../../base:protobuf_utils" ] | |
| 1206 } | |
| 1207 | |
| 1193 if (!build_with_chromium && is_clang) { | 1208 if (!build_with_chromium && is_clang) { |
| 1194 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 1209 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| 1195 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 1210 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| 1196 } | 1211 } |
| 1197 } | 1212 } |
| 1198 | 1213 |
| 1199 rtc_source_set("acm_receive_test") { | 1214 rtc_source_set("acm_receive_test") { |
| 1200 testonly = true | 1215 testonly = true |
| 1201 sources = [ | 1216 sources = [ |
| 1202 "acm2/acm_receive_test.cc", | 1217 "acm2/acm_receive_test.cc", |
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| 1322 ":isac", | 1337 ":isac", |
| 1323 ":isac_fix", | 1338 ":isac_fix", |
| 1324 ":neteq", | 1339 ":neteq", |
| 1325 ":neteq_unittest_tools", | 1340 ":neteq_unittest_tools", |
| 1326 "../../api/audio_codecs:audio_codecs_api", | 1341 "../../api/audio_codecs:audio_codecs_api", |
| 1327 "../../common_audio", | 1342 "../../common_audio", |
| 1328 "../../test:test_main", | 1343 "../../test:test_main", |
| 1329 "//testing/gtest", | 1344 "//testing/gtest", |
| 1330 ] | 1345 ] |
| 1331 | 1346 |
| 1347 if (rtc_enable_protobuf) { | |
| 1348 deps += [ "../../base:protobuf_utils" ] | |
| 1349 } | |
| 1350 | |
| 1332 data = audio_decoder_unittests_resources | 1351 data = audio_decoder_unittests_resources |
| 1333 | 1352 |
| 1334 if (is_android) { | 1353 if (is_android) { |
| 1335 deps += [ "//testing/android/native_test:native_test_native_code" ] | 1354 deps += [ "//testing/android/native_test:native_test_native_code" ] |
| 1336 shard_timeout = 900 | 1355 shard_timeout = 900 |
| 1337 } | 1356 } |
| 1338 if (is_ios) { | 1357 if (is_ios) { |
| 1339 deps += [ ":audio_decoder_unittests_bundle_data" ] | 1358 deps += [ ":audio_decoder_unittests_bundle_data" ] |
| 1340 } | 1359 } |
| 1341 } # audio_decoder_unittests | 1360 } # audio_decoder_unittests |
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| 2095 defines = audio_coding_defines | 2114 defines = audio_coding_defines |
| 2096 | 2115 |
| 2097 if (rtc_enable_protobuf) { | 2116 if (rtc_enable_protobuf) { |
| 2098 defines += [ | 2117 defines += [ |
| 2099 "WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP", | 2118 "WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP", |
| 2100 "WEBRTC_NETEQ_UNITTEST_BITEXACT", | 2119 "WEBRTC_NETEQ_UNITTEST_BITEXACT", |
| 2101 ] | 2120 ] |
| 2102 deps += [ | 2121 deps += [ |
| 2103 ":ana_config_proto", | 2122 ":ana_config_proto", |
| 2104 ":neteq_unittest_proto", | 2123 ":neteq_unittest_proto", |
| 2124 "../../base:protobuf_utils", | |
| 2105 ] | 2125 ] |
| 2106 } | 2126 } |
| 2107 | 2127 |
| 2108 if (!build_with_chromium && is_clang) { | 2128 if (!build_with_chromium && is_clang) { |
| 2109 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 2129 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| 2110 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 2130 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| 2111 } | 2131 } |
| 2112 } | 2132 } |
| 2113 } | 2133 } |
| 2114 | 2134 |
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| 2128 # webrtc/api/audio_codecs:builtin_audio_decoder_factory instead. | 2148 # webrtc/api/audio_codecs:builtin_audio_decoder_factory instead. |
| 2129 # TODO(kwiberg): Remove this. | 2149 # TODO(kwiberg): Remove this. |
| 2130 rtc_source_set("builtin_audio_decoder_factory") { | 2150 rtc_source_set("builtin_audio_decoder_factory") { |
| 2131 sources = [ | 2151 sources = [ |
| 2132 "codecs/builtin_audio_decoder_factory.h", | 2152 "codecs/builtin_audio_decoder_factory.h", |
| 2133 ] | 2153 ] |
| 2134 deps = [ | 2154 deps = [ |
| 2135 "../../api/audio_codecs:builtin_audio_decoder_factory", | 2155 "../../api/audio_codecs:builtin_audio_decoder_factory", |
| 2136 ] | 2156 ] |
| 2137 } | 2157 } |
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