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

Issue 2474913002: Logging basic bad call detection (Closed)
Patch Set: Comments Created 4 years 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 10
11 rtc_static_library("video") { 11 rtc_static_library("video") {
12 sources = [ 12 sources = [
13 "call_stats.cc", 13 "call_stats.cc",
14 "call_stats.h", 14 "call_stats.h",
15 "encoder_rtcp_feedback.cc", 15 "encoder_rtcp_feedback.cc",
16 "encoder_rtcp_feedback.h", 16 "encoder_rtcp_feedback.h",
17 "overuse_frame_detector.cc", 17 "overuse_frame_detector.cc",
18 "overuse_frame_detector.h", 18 "overuse_frame_detector.h",
19 "payload_router.cc", 19 "payload_router.cc",
20 "payload_router.h", 20 "payload_router.h",
21 "quality_threshold.cc",
22 "quality_threshold.h",
21 "receive_statistics_proxy.cc", 23 "receive_statistics_proxy.cc",
22 "receive_statistics_proxy.h", 24 "receive_statistics_proxy.h",
23 "report_block_stats.cc", 25 "report_block_stats.cc",
24 "report_block_stats.h", 26 "report_block_stats.h",
25 "rtp_stream_receiver.cc", 27 "rtp_stream_receiver.cc",
26 "rtp_stream_receiver.h", 28 "rtp_stream_receiver.h",
27 "rtp_streams_synchronizer.cc", 29 "rtp_streams_synchronizer.cc",
28 "rtp_streams_synchronizer.h", 30 "rtp_streams_synchronizer.h",
29 "send_delay_stats.cc", 31 "send_delay_stats.cc",
30 "send_delay_stats.h", 32 "send_delay_stats.h",
(...skipping 112 matching lines...) Expand 10 before | Expand all | Expand 10 after
143 145
144 # TODO(pbos): Rename test suite. 146 # TODO(pbos): Rename test suite.
145 rtc_source_set("video_tests") { 147 rtc_source_set("video_tests") {
146 testonly = true 148 testonly = true
147 sources = [ 149 sources = [
148 "call_stats_unittest.cc", 150 "call_stats_unittest.cc",
149 "encoder_rtcp_feedback_unittest.cc", 151 "encoder_rtcp_feedback_unittest.cc",
150 "end_to_end_tests.cc", 152 "end_to_end_tests.cc",
151 "overuse_frame_detector_unittest.cc", 153 "overuse_frame_detector_unittest.cc",
152 "payload_router_unittest.cc", 154 "payload_router_unittest.cc",
155 "quality_threshold_unittest.cc",
153 "receive_statistics_proxy_unittest.cc", 156 "receive_statistics_proxy_unittest.cc",
154 "report_block_stats_unittest.cc", 157 "report_block_stats_unittest.cc",
155 "send_delay_stats_unittest.cc", 158 "send_delay_stats_unittest.cc",
156 "send_statistics_proxy_unittest.cc", 159 "send_statistics_proxy_unittest.cc",
157 "stats_counter_unittest.cc", 160 "stats_counter_unittest.cc",
158 "stream_synchronization_unittest.cc", 161 "stream_synchronization_unittest.cc",
159 "video_send_stream_tests.cc", 162 "video_send_stream_tests.cc",
160 "vie_encoder_unittest.cc", 163 "vie_encoder_unittest.cc",
161 "vie_remb_unittest.cc", 164 "vie_remb_unittest.cc",
162 ] 165 ]
163 deps = [ 166 deps = [
164 ":video", 167 ":video",
165 "../media:rtc_media_base", 168 "../media:rtc_media_base",
166 "//testing/gmock", 169 "//testing/gmock",
167 "//testing/gtest", 170 "//testing/gtest",
168 ] 171 ]
169 if (!build_with_chromium && is_clang) { 172 if (!build_with_chromium && is_clang) {
170 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). 173 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
171 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] 174 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
172 } 175 }
173 if (rtc_use_h264) { 176 if (rtc_use_h264) {
174 defines = [ "WEBRTC_USE_H264" ] 177 defines = [ "WEBRTC_USE_H264" ]
175 } 178 }
176 } 179 }
177 } 180 }
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