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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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1096 ":neteq", | 1096 ":neteq", |
1097 "..:module_api", | 1097 "..:module_api", |
1098 "../..:webrtc_common", | 1098 "../..:webrtc_common", |
1099 "../../api/audio_codecs:audio_codecs_api", | 1099 "../../api/audio_codecs:audio_codecs_api", |
1100 "../../api/audio_codecs:builtin_audio_decoder_factory", | 1100 "../../api/audio_codecs:builtin_audio_decoder_factory", |
1101 "../../base:rtc_base_approved", | 1101 "../../base:rtc_base_approved", |
1102 "../rtp_rtcp", | 1102 "../rtp_rtcp", |
1103 ] | 1103 ] |
1104 } | 1104 } |
1105 | 1105 |
1106 rtc_source_set("neteq_test_tools") { | |
1107 testonly = true | |
1108 sources = [ | |
1109 "neteq/tools/audio_checksum.h", | |
1110 "neteq/tools/audio_loop.cc", | |
1111 "neteq/tools/audio_loop.h", | |
1112 "neteq/tools/constant_pcm_packet_source.cc", | |
1113 "neteq/tools/constant_pcm_packet_source.h", | |
1114 "neteq/tools/output_audio_file.h", | |
1115 "neteq/tools/output_wav_file.h", | |
1116 "neteq/tools/rtp_file_source.cc", | |
1117 "neteq/tools/rtp_file_source.h", | |
1118 "neteq/tools/rtp_generator.cc", | |
1119 "neteq/tools/rtp_generator.h", | |
1120 ] | |
1121 | |
1122 public_configs = [ ":neteq_tools_config" ] | |
1123 | |
1124 if (!build_with_chromium && is_clang) { | |
1125 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | |
1126 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | |
1127 } | |
1128 | |
1129 deps = [ | |
1130 ":pcm16b", | |
1131 "..:module_api", | |
1132 "../..:webrtc_common", | |
1133 "../../base:rtc_base_approved", | |
1134 "../../base:rtc_base_tests_utils", | |
1135 "../../common_audio", | |
1136 "../../test:rtp_test_utils", | |
1137 "../rtp_rtcp", | |
1138 ] | |
1139 | |
1140 public_deps = [ | |
1141 ":neteq_tools", | |
1142 ":neteq_tools_minimal", | |
1143 ] | |
1144 | |
1145 if (rtc_enable_protobuf) { | |
1146 sources += [ | |
1147 "neteq/tools/neteq_packet_source_input.cc", | |
1148 "neteq/tools/neteq_packet_source_input.h", | |
1149 ] | |
1150 deps += [ ":rtc_event_log_source" ] | |
1151 } | |
1152 } | |
1153 | |
1154 config("neteq_tools_config") { | |
1155 include_dirs = [ "tools" ] | |
1156 } | |
1157 | |
1158 rtc_source_set("neteq_tools") { | |
1159 sources = [ | |
1160 "neteq/tools/fake_decode_from_file.cc", | |
1161 "neteq/tools/fake_decode_from_file.h", | |
1162 "neteq/tools/input_audio_file.cc", | |
1163 "neteq/tools/input_audio_file.h", | |
1164 "neteq/tools/neteq_replacement_input.cc", | |
1165 "neteq/tools/neteq_replacement_input.h", | |
1166 "neteq/tools/resample_input_audio_file.cc", | |
1167 "neteq/tools/resample_input_audio_file.h", | |
1168 ] | |
1169 | |
1170 public_configs = [ ":neteq_tools_config" ] | |
1171 | |
1172 if (!build_with_chromium && is_clang) { | |
1173 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | |
1174 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | |
1175 } | |
1176 | |
1177 deps = [ | |
1178 "../..:webrtc_common", | |
1179 "../../api/audio_codecs:audio_codecs_api", | |
1180 "../../base:rtc_base_approved", | |
1181 "../../common_audio", | |
1182 "../rtp_rtcp", | |
1183 ] | |
1184 | |
1185 public_deps = [ | |
1186 ":neteq_tools_minimal", | |
1187 ] | |
1188 } | |
1189 | |
1190 if (rtc_enable_protobuf) { | |
1191 rtc_static_library("rtc_event_log_source") { | |
1192 testonly = true | |
1193 | |
1194 # TODO(kjellander): Remove (bugs.webrtc.org/6828) | |
1195 # Needs call.h to be moved to webrtc/api first. | |
1196 check_includes = false | |
1197 | |
1198 sources = [ | |
1199 "neteq/tools/rtc_event_log_source.cc", | |
1200 "neteq/tools/rtc_event_log_source.h", | |
1201 ] | |
1202 | |
1203 if (!build_with_chromium && is_clang) { | |
1204 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | |
1205 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | |
1206 } | |
1207 | |
1208 deps = [ | |
1209 "../../base:rtc_base_approved", | |
1210 "../../logging:rtc_event_log_parser", | |
1211 ] | |
1212 public_deps = [ | |
1213 "../../logging:rtc_event_log_proto", | |
1214 ] | |
1215 } | |
1216 } | |
1217 | |
1218 if (rtc_include_tests) { | 1106 if (rtc_include_tests) { |
1219 group("audio_coding_tests") { | 1107 group("audio_coding_tests") { |
1220 testonly = true | 1108 testonly = true |
1221 public_deps = [ | 1109 public_deps = [ |
1222 ":RTPchange", | 1110 ":RTPchange", |
1223 ":RTPencode", | 1111 ":RTPencode", |
1224 ":RTPjitter", | 1112 ":RTPjitter", |
1225 ":RTPtimeshift", | 1113 ":RTPtimeshift", |
1226 ":acm_receive_test", | 1114 ":acm_receive_test", |
1227 ":acm_send_test", | 1115 ":acm_send_test", |
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1500 } # audio_decoder_unittests | 1388 } # audio_decoder_unittests |
1501 | 1389 |
1502 if (rtc_enable_protobuf) { | 1390 if (rtc_enable_protobuf) { |
1503 proto_library("neteq_unittest_proto") { | 1391 proto_library("neteq_unittest_proto") { |
1504 sources = [ | 1392 sources = [ |
1505 "neteq/neteq_unittest.proto", | 1393 "neteq/neteq_unittest.proto", |
1506 ] | 1394 ] |
1507 proto_out_dir = "webrtc/modules/audio_coding/neteq" | 1395 proto_out_dir = "webrtc/modules/audio_coding/neteq" |
1508 } | 1396 } |
1509 | 1397 |
| 1398 rtc_static_library("rtc_event_log_source") { |
| 1399 testonly = true |
| 1400 |
| 1401 # TODO(kjellander): Remove (bugs.webrtc.org/6828) |
| 1402 # Needs call.h to be moved to webrtc/api first. |
| 1403 check_includes = false |
| 1404 |
| 1405 sources = [ |
| 1406 "neteq/tools/rtc_event_log_source.cc", |
| 1407 "neteq/tools/rtc_event_log_source.h", |
| 1408 ] |
| 1409 |
| 1410 if (!build_with_chromium && is_clang) { |
| 1411 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163)
. |
| 1412 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| 1413 } |
| 1414 |
| 1415 deps = [ |
| 1416 "../../base:rtc_base_approved", |
| 1417 "../../logging:rtc_event_log_parser", |
| 1418 ] |
| 1419 public_deps = [ |
| 1420 "../../logging:rtc_event_log_proto", |
| 1421 ] |
| 1422 } |
| 1423 |
1510 rtc_test("neteq_rtpplay") { | 1424 rtc_test("neteq_rtpplay") { |
1511 testonly = true | 1425 testonly = true |
1512 defines = [] | 1426 defines = [] |
1513 deps = [ | 1427 deps = [ |
1514 "..:module_api", | 1428 "..:module_api", |
1515 ] | 1429 ] |
1516 sources = [ | 1430 sources = [ |
1517 "neteq/tools/neteq_rtpplay.cc", | 1431 "neteq/tools/neteq_rtpplay.cc", |
1518 ] | 1432 ] |
1519 | 1433 |
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1621 "..:module_api", | 1535 "..:module_api", |
1622 "../..:webrtc_common", | 1536 "../..:webrtc_common", |
1623 "../../api/audio_codecs:builtin_audio_decoder_factory", | 1537 "../../api/audio_codecs:builtin_audio_decoder_factory", |
1624 "../../base:rtc_base_approved", | 1538 "../../base:rtc_base_approved", |
1625 "../../test:test_support", | 1539 "../../test:test_support", |
1626 "//testing/gtest", | 1540 "//testing/gtest", |
1627 "//third_party/gflags", | 1541 "//third_party/gflags", |
1628 ] | 1542 ] |
1629 } | 1543 } |
1630 | 1544 |
| 1545 config("neteq_tools_config") { |
| 1546 include_dirs = [ "tools" ] |
| 1547 } |
| 1548 |
| 1549 rtc_source_set("neteq_tools") { |
| 1550 sources = [ |
| 1551 "neteq/tools/fake_decode_from_file.cc", |
| 1552 "neteq/tools/fake_decode_from_file.h", |
| 1553 "neteq/tools/input_audio_file.cc", |
| 1554 "neteq/tools/input_audio_file.h", |
| 1555 "neteq/tools/neteq_replacement_input.cc", |
| 1556 "neteq/tools/neteq_replacement_input.h", |
| 1557 "neteq/tools/resample_input_audio_file.cc", |
| 1558 "neteq/tools/resample_input_audio_file.h", |
| 1559 ] |
| 1560 |
| 1561 public_configs = [ ":neteq_tools_config" ] |
| 1562 |
| 1563 if (!build_with_chromium && is_clang) { |
| 1564 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| 1565 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| 1566 } |
| 1567 |
| 1568 deps = [ |
| 1569 "../..:webrtc_common", |
| 1570 "../../api/audio_codecs:audio_codecs_api", |
| 1571 "../../base:rtc_base_approved", |
| 1572 "../../common_audio", |
| 1573 "../rtp_rtcp", |
| 1574 ] |
| 1575 |
| 1576 public_deps = [ |
| 1577 ":neteq_tools_minimal", |
| 1578 ] |
| 1579 } |
| 1580 |
| 1581 rtc_source_set("neteq_test_tools") { |
| 1582 testonly = true |
| 1583 sources = [ |
| 1584 "neteq/tools/audio_checksum.h", |
| 1585 "neteq/tools/audio_loop.cc", |
| 1586 "neteq/tools/audio_loop.h", |
| 1587 "neteq/tools/constant_pcm_packet_source.cc", |
| 1588 "neteq/tools/constant_pcm_packet_source.h", |
| 1589 "neteq/tools/output_audio_file.h", |
| 1590 "neteq/tools/output_wav_file.h", |
| 1591 "neteq/tools/rtp_file_source.cc", |
| 1592 "neteq/tools/rtp_file_source.h", |
| 1593 "neteq/tools/rtp_generator.cc", |
| 1594 "neteq/tools/rtp_generator.h", |
| 1595 ] |
| 1596 |
| 1597 public_configs = [ ":neteq_tools_config" ] |
| 1598 |
| 1599 if (!build_with_chromium && is_clang) { |
| 1600 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| 1601 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| 1602 } |
| 1603 |
| 1604 deps = [ |
| 1605 ":pcm16b", |
| 1606 "..:module_api", |
| 1607 "../..:webrtc_common", |
| 1608 "../../base:rtc_base_approved", |
| 1609 "../../base:rtc_base_tests_utils", |
| 1610 "../../common_audio", |
| 1611 "../../test:rtp_test_utils", |
| 1612 "../rtp_rtcp", |
| 1613 ] |
| 1614 |
| 1615 public_deps = [ |
| 1616 ":neteq_tools", |
| 1617 ":neteq_tools_minimal", |
| 1618 ] |
| 1619 |
| 1620 if (rtc_enable_protobuf) { |
| 1621 sources += [ |
| 1622 "neteq/tools/neteq_packet_source_input.cc", |
| 1623 "neteq/tools/neteq_packet_source_input.h", |
| 1624 ] |
| 1625 deps += [ ":rtc_event_log_source" ] |
| 1626 } |
| 1627 } |
| 1628 |
1631 rtc_source_set("neteq_test_tools_deprecated") { | 1629 rtc_source_set("neteq_test_tools_deprecated") { |
1632 testonly = true | 1630 testonly = true |
1633 sources = [ | 1631 sources = [ |
1634 "neteq/test/NETEQTEST_DummyRTPpacket.cc", | 1632 "neteq/test/NETEQTEST_DummyRTPpacket.cc", |
1635 "neteq/test/NETEQTEST_DummyRTPpacket.h", | 1633 "neteq/test/NETEQTEST_DummyRTPpacket.h", |
1636 "neteq/test/NETEQTEST_RTPpacket.cc", | 1634 "neteq/test/NETEQTEST_RTPpacket.cc", |
1637 "neteq/test/NETEQTEST_RTPpacket.h", | 1635 "neteq/test/NETEQTEST_RTPpacket.h", |
1638 ] | 1636 ] |
1639 | 1637 |
1640 deps = [ | 1638 deps = [ |
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2235 # webrtc/api/audio_codecs:builtin_audio_decoder_factory instead. | 2233 # webrtc/api/audio_codecs:builtin_audio_decoder_factory instead. |
2236 # TODO(ossu): Remove this. | 2234 # TODO(ossu): Remove this. |
2237 rtc_source_set("builtin_audio_encoder_factory") { | 2235 rtc_source_set("builtin_audio_encoder_factory") { |
2238 sources = [ | 2236 sources = [ |
2239 "codecs/builtin_audio_encoder_factory.h", | 2237 "codecs/builtin_audio_encoder_factory.h", |
2240 ] | 2238 ] |
2241 deps = [ | 2239 deps = [ |
2242 "../../api/audio_codecs:builtin_audio_encoder_factory", | 2240 "../../api/audio_codecs:builtin_audio_encoder_factory", |
2243 ] | 2241 ] |
2244 } | 2242 } |
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