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Side by Side Diff: webrtc/video/BUILD.gn

Issue 2064523002: GN: Add video_engine_tests (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: audio_receive_stream_unittest compile Created 4 years, 6 months ago
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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 import("//testing/test.gni")
10 11
11 source_set("video") { 12 source_set("video") {
12 sources = [ 13 sources = [
13 "call_stats.cc", 14 "call_stats.cc",
14 "call_stats.h", 15 "call_stats.h",
15 "encoded_frame_callback_adapter.cc", 16 "encoded_frame_callback_adapter.cc",
16 "encoded_frame_callback_adapter.h", 17 "encoded_frame_callback_adapter.h",
17 "encoder_state_feedback.cc", 18 "encoder_state_feedback.cc",
18 "encoder_state_feedback.h", 19 "encoder_state_feedback.h",
19 "overuse_frame_detector.cc", 20 "overuse_frame_detector.cc",
(...skipping 52 matching lines...) Expand 10 before | Expand all | Expand 10 after
72 "../modules/remote_bitrate_estimator", 73 "../modules/remote_bitrate_estimator",
73 "../modules/rtp_rtcp", 74 "../modules/rtp_rtcp",
74 "../modules/utility", 75 "../modules/utility",
75 "../modules/video_capture:video_capture_module", 76 "../modules/video_capture:video_capture_module",
76 "../modules/video_coding", 77 "../modules/video_coding",
77 "../modules/video_processing", 78 "../modules/video_processing",
78 "../system_wrappers", 79 "../system_wrappers",
79 "../voice_engine", 80 "../voice_engine",
80 ] 81 ]
81 } 82 }
83
84 if (rtc_include_tests) {
85 # TODO(pbos): Rename test suite.
86 source_set("video_tests") {
87 testonly = true
88 sources = [
89 "call_stats_unittest.cc",
90 "encoder_state_feedback_unittest.cc",
91 "end_to_end_tests.cc",
92 "overuse_frame_detector_unittest.cc",
93 "payload_router_unittest.cc",
94 "report_block_stats_unittest.cc",
95 "send_delay_stats_unittest.cc",
96 "send_statistics_proxy_unittest.cc",
97 "stats_counter_unittest.cc",
98 "stream_synchronization_unittest.cc",
99 "video_capture_input_unittest.cc",
100 "video_decoder_unittest.cc",
101 "video_encoder_unittest.cc",
102 "video_send_stream_tests.cc",
103 "vie_remb_unittest.cc",
104 ]
105 deps = [
106 ":video",
107 "//testing/gmock",
108 "//testing/gtest",
109 ]
110 if (is_clang) {
111 # Suppress warnings from the Chromium Clang plugin.
112 # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
113 configs -= [ "//build/config/clang:find_bad_constructs" ]
114 }
115 }
116 }
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