Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(351)

Side by Side Diff: webrtc/modules/audio_coding/BUILD.gn

Issue 2846493002: Reland of Enable GN check for webrtc/base (Closed)
Patch Set: Running chromium trybots again :) Created 3 years, 8 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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")
(...skipping 1115 matching lines...) Expand 10 before | Expand all | Expand 10 after
1126 1126
1127 deps = [ 1127 deps = [
1128 ":audio_encoder_interface", 1128 ":audio_encoder_interface",
1129 ":neteq", 1129 ":neteq",
1130 "../..:webrtc_common", 1130 "../..:webrtc_common",
1131 "../../api/audio_codecs:builtin_audio_decoder_factory", 1131 "../../api/audio_codecs:builtin_audio_decoder_factory",
1132 "../../base:rtc_base_approved", 1132 "../../base:rtc_base_approved",
1133 ] 1133 ]
1134 } 1134 }
1135 1135
1136 config("neteq_unittest_tools_config") {
1137 include_dirs = [ "tools" ]
1138 }
1139
1140 rtc_source_set("neteq_unittest_tools") {
1141 testonly = true
1142 sources = [
1143 "neteq/tools/audio_checksum.h",
1144 "neteq/tools/audio_loop.cc",
1145 "neteq/tools/audio_loop.h",
1146 "neteq/tools/audio_sink.cc",
1147 "neteq/tools/audio_sink.h",
1148 "neteq/tools/constant_pcm_packet_source.cc",
1149 "neteq/tools/constant_pcm_packet_source.h",
1150 "neteq/tools/fake_decode_from_file.cc",
1151 "neteq/tools/fake_decode_from_file.h",
1152 "neteq/tools/input_audio_file.cc",
1153 "neteq/tools/input_audio_file.h",
1154 "neteq/tools/neteq_input.h",
1155 "neteq/tools/neteq_replacement_input.cc",
1156 "neteq/tools/neteq_replacement_input.h",
1157 "neteq/tools/output_audio_file.h",
1158 "neteq/tools/output_wav_file.h",
1159 "neteq/tools/packet.cc",
1160 "neteq/tools/packet.h",
1161 "neteq/tools/packet_source.cc",
1162 "neteq/tools/packet_source.h",
1163 "neteq/tools/resample_input_audio_file.cc",
1164 "neteq/tools/resample_input_audio_file.h",
1165 "neteq/tools/rtp_file_source.cc",
1166 "neteq/tools/rtp_file_source.h",
1167 "neteq/tools/rtp_generator.cc",
1168 "neteq/tools/rtp_generator.h",
1169 ]
1170
1171 public_configs = [ ":neteq_unittest_tools_config" ]
1172
1173 if (!build_with_chromium && is_clang) {
1174 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
1175 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
1176 }
1177
1178 deps = [
1179 ":audio_encoder_interface",
1180 ":pcm16b",
1181 "..:module_api",
1182 "../..:webrtc_common",
1183 "../../api/audio_codecs:audio_codecs_api",
1184 "../../base:rtc_base_approved",
1185 "../../base:rtc_base_tests_utils",
1186 "../../common_audio",
1187 "../../test:rtp_test_utils",
1188 "../rtp_rtcp",
1189 ]
1190
1191 public_deps = [
1192 ":neteq_test_minimal",
1193 ]
1194
1195 if (rtc_enable_protobuf) {
1196 sources += [
1197 "neteq/tools/neteq_packet_source_input.cc",
1198 "neteq/tools/neteq_packet_source_input.h",
1199 ]
1200 deps += [ ":rtc_event_log_source" ]
1201 }
1202 }
1203
1204 if (rtc_enable_protobuf) {
1205 rtc_static_library("rtc_event_log_source") {
1206 testonly = true
1207
1208 # TODO(kjellander): Remove (bugs.webrtc.org/6828)
1209 # Needs call.h to be moved to webrtc/api first.
1210 check_includes = false
1211
1212 sources = [
1213 "neteq/tools/rtc_event_log_source.cc",
1214 "neteq/tools/rtc_event_log_source.h",
1215 ]
1216
1217 if (!build_with_chromium && is_clang) {
1218 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
1219 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
1220 }
1221
1222 deps = [
1223 "../../base:rtc_base_approved",
1224 "../../logging:rtc_event_log_parser",
1225 ]
1226 public_deps = [
1227 "../../logging:rtc_event_log_proto",
1228 ]
1229 }
1230 }
1231
1136 if (rtc_include_tests) { 1232 if (rtc_include_tests) {
1137 group("audio_coding_tests") { 1233 group("audio_coding_tests") {
1138 testonly = true 1234 testonly = true
1139 public_deps = [ 1235 public_deps = [
1140 ":RTPchange", 1236 ":RTPchange",
1141 ":RTPencode", 1237 ":RTPencode",
1142 ":RTPjitter", 1238 ":RTPjitter",
1143 ":RTPtimeshift", 1239 ":RTPtimeshift",
1144 ":acm_receive_test", 1240 ":acm_receive_test",
1145 ":acm_send_test", 1241 ":acm_send_test",
(...skipping 253 matching lines...) Expand 10 before | Expand all | Expand 10 after
1399 } # audio_decoder_unittests 1495 } # audio_decoder_unittests
1400 1496
1401 if (rtc_enable_protobuf) { 1497 if (rtc_enable_protobuf) {
1402 proto_library("neteq_unittest_proto") { 1498 proto_library("neteq_unittest_proto") {
1403 sources = [ 1499 sources = [
1404 "neteq/neteq_unittest.proto", 1500 "neteq/neteq_unittest.proto",
1405 ] 1501 ]
1406 proto_out_dir = "webrtc/modules/audio_coding/neteq" 1502 proto_out_dir = "webrtc/modules/audio_coding/neteq"
1407 } 1503 }
1408 1504
1409 rtc_static_library("rtc_event_log_source") {
1410 testonly = true
1411
1412 # TODO(kjellander): Remove (bugs.webrtc.org/6828)
1413 # Needs call.h to be moved to webrtc/api first.
1414 check_includes = false
1415
1416 sources = [
1417 "neteq/tools/rtc_event_log_source.cc",
1418 "neteq/tools/rtc_event_log_source.h",
1419 ]
1420
1421 if (!build_with_chromium && is_clang) {
1422 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163) .
1423 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
1424 }
1425
1426 deps = [
1427 "../../base:rtc_base_approved",
1428 "../../logging:rtc_event_log_parser",
1429 ]
1430 public_deps = [
1431 "../../logging:rtc_event_log_proto",
1432 ]
1433 }
1434
1435 rtc_test("neteq_rtpplay") { 1505 rtc_test("neteq_rtpplay") {
1436 testonly = true 1506 testonly = true
1437 defines = [] 1507 defines = []
1438 deps = [ 1508 deps = [
1439 "..:module_api", 1509 "..:module_api",
1440 ] 1510 ]
1441 sources = [ 1511 sources = [
1442 "neteq/tools/neteq_rtpplay.cc", 1512 "neteq/tools/neteq_rtpplay.cc",
1443 ] 1513 ]
1444 1514
(...skipping 101 matching lines...) Expand 10 before | Expand all | Expand 10 after
1546 "..:module_api", 1616 "..:module_api",
1547 "../..:webrtc_common", 1617 "../..:webrtc_common",
1548 "../../api/audio_codecs:builtin_audio_decoder_factory", 1618 "../../api/audio_codecs:builtin_audio_decoder_factory",
1549 "../../base:rtc_base_approved", 1619 "../../base:rtc_base_approved",
1550 "../../test:test_support", 1620 "../../test:test_support",
1551 "//testing/gtest", 1621 "//testing/gtest",
1552 "//third_party/gflags", 1622 "//third_party/gflags",
1553 ] 1623 ]
1554 } 1624 }
1555 1625
1556 config("neteq_unittest_tools_config") {
1557 include_dirs = [ "tools" ]
1558 }
1559
1560 rtc_source_set("neteq_unittest_tools") {
1561 testonly = true
1562 sources = [
1563 "neteq/tools/audio_checksum.h",
1564 "neteq/tools/audio_loop.cc",
1565 "neteq/tools/audio_loop.h",
1566 "neteq/tools/audio_sink.cc",
1567 "neteq/tools/audio_sink.h",
1568 "neteq/tools/constant_pcm_packet_source.cc",
1569 "neteq/tools/constant_pcm_packet_source.h",
1570 "neteq/tools/fake_decode_from_file.cc",
1571 "neteq/tools/fake_decode_from_file.h",
1572 "neteq/tools/input_audio_file.cc",
1573 "neteq/tools/input_audio_file.h",
1574 "neteq/tools/neteq_input.h",
1575 "neteq/tools/neteq_replacement_input.cc",
1576 "neteq/tools/neteq_replacement_input.h",
1577 "neteq/tools/output_audio_file.h",
1578 "neteq/tools/output_wav_file.h",
1579 "neteq/tools/packet.cc",
1580 "neteq/tools/packet.h",
1581 "neteq/tools/packet_source.cc",
1582 "neteq/tools/packet_source.h",
1583 "neteq/tools/resample_input_audio_file.cc",
1584 "neteq/tools/resample_input_audio_file.h",
1585 "neteq/tools/rtp_file_source.cc",
1586 "neteq/tools/rtp_file_source.h",
1587 "neteq/tools/rtp_generator.cc",
1588 "neteq/tools/rtp_generator.h",
1589 ]
1590
1591 public_configs = [ ":neteq_unittest_tools_config" ]
1592
1593 if (!build_with_chromium && is_clang) {
1594 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
1595 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
1596 }
1597
1598 deps = [
1599 ":audio_encoder_interface",
1600 ":pcm16b",
1601 "..:module_api",
1602 "../..:webrtc_common",
1603 "../../api/audio_codecs:audio_codecs_api",
1604 "../../base:rtc_base_approved",
1605 "../../common_audio",
1606 "../../test:rtp_test_utils",
1607 "../rtp_rtcp",
1608 ]
1609
1610 public_deps = [
1611 ":neteq_test_minimal",
1612 ]
1613
1614 if (rtc_enable_protobuf) {
1615 sources += [
1616 "neteq/tools/neteq_packet_source_input.cc",
1617 "neteq/tools/neteq_packet_source_input.h",
1618 ]
1619 deps += [ ":rtc_event_log_source" ]
1620 }
1621 }
1622
1623 rtc_source_set("neteq_test_tools") { 1626 rtc_source_set("neteq_test_tools") {
1624 testonly = true 1627 testonly = true
1625 sources = [ 1628 sources = [
1626 "neteq/test/NETEQTEST_DummyRTPpacket.cc", 1629 "neteq/test/NETEQTEST_DummyRTPpacket.cc",
1627 "neteq/test/NETEQTEST_DummyRTPpacket.h", 1630 "neteq/test/NETEQTEST_DummyRTPpacket.h",
1628 "neteq/test/NETEQTEST_RTPpacket.cc", 1631 "neteq/test/NETEQTEST_RTPpacket.cc",
1629 "neteq/test/NETEQTEST_RTPpacket.h", 1632 "neteq/test/NETEQTEST_RTPpacket.h",
1630 ] 1633 ]
1631 1634
1632 deps = [ 1635 deps = [
(...skipping 571 matching lines...) Expand 10 before | Expand all | Expand 10 after
2204 # webrtc/api/audio_codecs:builtin_audio_decoder_factory instead. 2207 # webrtc/api/audio_codecs:builtin_audio_decoder_factory instead.
2205 # TODO(kwiberg): Remove this. 2208 # TODO(kwiberg): Remove this.
2206 rtc_source_set("builtin_audio_decoder_factory") { 2209 rtc_source_set("builtin_audio_decoder_factory") {
2207 sources = [ 2210 sources = [
2208 "codecs/builtin_audio_decoder_factory.h", 2211 "codecs/builtin_audio_decoder_factory.h",
2209 ] 2212 ]
2210 deps = [ 2213 deps = [
2211 "../../api/audio_codecs:builtin_audio_decoder_factory", 2214 "../../api/audio_codecs:builtin_audio_decoder_factory",
2212 ] 2215 ]
2213 } 2216 }
OLDNEW
« no previous file with comments | « webrtc/common_audio/resampler/sinc_resampler_unittest.cc ('k') | webrtc/system_wrappers/BUILD.gn » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698