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1 # Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. | 1 # Copyright (c) 2016 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 import("audio_coding/audio_coding.gni") | 10 import("audio_coding/audio_coding.gni") |
11 import("video_coding/video_coding.gni") | |
11 import("//testing/test.gni") | 12 import("//testing/test.gni") |
12 | 13 |
13 declare_args() { | 14 declare_args() { |
14 # Desktop capturer is supported only on Windows, OSX and Linux. | 15 # Desktop capturer is supported only on Windows, OSX and Linux. |
15 rtc_desktop_capture_supported = is_win || is_mac || is_linux | 16 rtc_desktop_capture_supported = is_win || is_mac || is_linux |
16 } | 17 } |
17 | 18 |
18 group("modules") { | 19 group("modules") { |
19 deps = [ | 20 deps = [ |
20 "audio_coding", | 21 "audio_coding", |
21 "audio_conference_mixer", | 22 "audio_conference_mixer", |
22 "audio_device", | 23 "audio_device", |
23 "audio_mixer", | 24 "audio_mixer", |
24 "audio_processing", | 25 "audio_processing", |
25 "bitrate_controller", | 26 "bitrate_controller", |
26 "desktop_capture", | 27 "desktop_capture", |
27 "media_file", | 28 "media_file", |
28 "rtp_rtcp", | 29 "rtp_rtcp", |
29 "utility", | 30 "utility", |
30 "video_coding", | 31 "video_coding", |
31 "video_processing", | 32 "video_processing", |
32 ] | 33 ] |
33 } | 34 } |
34 | 35 |
35 if (rtc_include_tests) { | 36 if (rtc_include_tests) { |
37 test("modules_tests") { | |
38 testonly = true | |
39 | |
40 configs += [ "..:common_config" ] | |
41 public_configs = [ "..:common_inherited_config" ] | |
42 | |
43 defines = audio_coding_defines + videoprocessor_defines | |
44 | |
45 deps = [ | |
46 "..:webrtc_common", | |
47 "../common_video", | |
48 "../modules/audio_coding", | |
49 "../modules/rtp_rtcp", | |
50 "../modules/utility", | |
51 "../modules/video_coding", | |
52 "../modules/video_coding:video_codecs_test_framework", | |
53 "../system_wrappers", | |
54 "../test:test_support", | |
55 "../test:test_support_main", | |
56 "//testing/gtest", | |
57 ] | |
58 | |
59 sources = [ | |
60 "audio_coding/test/APITest.cc", | |
61 "audio_coding/test/Channel.cc", | |
62 "audio_coding/test/EncodeDecodeTest.cc", | |
63 "audio_coding/test/PCMFile.cc", | |
64 "audio_coding/test/PacketLossTest.cc", | |
65 "audio_coding/test/RTPFile.cc", | |
66 "audio_coding/test/TestAllCodecs.cc", | |
67 "audio_coding/test/TestRedFec.cc", | |
68 "audio_coding/test/TestStereo.cc", | |
69 "audio_coding/test/TestVADDTX.cc", | |
70 "audio_coding/test/Tester.cc", | |
71 "audio_coding/test/TwoWayCommunication.cc", | |
72 "audio_coding/test/iSACTest.cc", | |
73 "audio_coding/test/opus_test.cc", | |
74 "audio_coding/test/target_delay_unittest.cc", | |
75 "audio_coding/test/utility.cc", | |
76 "rtp_rtcp/test/testFec/test_fec.cc", | |
77 "video_coding/codecs/test/videoprocessor_integrationtest.cc", | |
78 "video_coding/codecs/vp8/test/vp8_impl_unittest.cc", | |
79 ] | |
80 | |
81 if (is_android) { | |
82 deps += [ "//testing/android/native_test:native_test_native_code" ] | |
83 } | |
84 if (is_ios) { | |
kjellander_webrtc
2016/08/03 15:22:49
Make this
if (is_android || is_ios) {
instead. I
ehmaldonado_webrtc
2016/08/03 15:40:43
Done.
| |
85 data = [ | |
86 "//resources/audio_coding/testfile32kHz.pcm", | |
87 "//resources/audio_coding/teststereo32kHz.pcm", | |
88 "//resources/foreman_cif.yuv", | |
89 "//resources/paris_qcif.yuv", | |
90 ] | |
91 } | |
92 | |
93 if (is_clang) { | |
94 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | |
95 configs -= [ "//build/config/clang:find_bad_constructs" ] | |
96 } | |
97 if (is_win) { | |
98 cflags = [ | |
99 # TODO(phoglund): get rid of 4373 supression when | |
100 # http://code.google.com/p/webrtc/issues/detail?id=261 is solved. | |
101 # legacy warning for ignoring const / volatile in signatures. | |
102 "/wd4373", | |
103 ] | |
104 } | |
105 } | |
106 | |
36 test("modules_unittests") { | 107 test("modules_unittests") { |
37 testonly = true | 108 testonly = true |
38 | 109 |
39 defines = audio_coding_defines | 110 defines = audio_coding_defines |
40 deps = [] | 111 deps = [] |
41 sources = [ | 112 sources = [ |
42 "audio_coding/acm2/acm_receiver_unittest_oldapi.cc", | 113 "audio_coding/acm2/acm_receiver_unittest_oldapi.cc", |
43 "audio_coding/acm2/audio_coding_module_unittest_oldapi.cc", | 114 "audio_coding/acm2/audio_coding_module_unittest_oldapi.cc", |
44 "audio_coding/acm2/call_statistics_unittest.cc", | 115 "audio_coding/acm2/call_statistics_unittest.cc", |
45 "audio_coding/acm2/codec_manager_unittest.cc", | 116 "audio_coding/acm2/codec_manager_unittest.cc", |
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535 "audio_device/ios/objc/RTCAudioSessionTest.mm", | 606 "audio_device/ios/objc/RTCAudioSessionTest.mm", |
536 "video_coding/codecs/h264/h264_video_toolbox_nalu_unittest.cc", | 607 "video_coding/codecs/h264/h264_video_toolbox_nalu_unittest.cc", |
537 ] | 608 ] |
538 | 609 |
539 ldflags = [ "-ObjC" ] | 610 ldflags = [ "-ObjC" ] |
540 | 611 |
541 # TODO(kjellander): Mac bundle files. | 612 # TODO(kjellander): Mac bundle files. |
542 } | 613 } |
543 } | 614 } |
544 } | 615 } |
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