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Issue 2510033004: Add rtc_use_memcheck flag, update MB and GN to handle it, and add gni files listing the runtime deps (Closed)
Patch Set: Fixed android and added unittest. 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/config/arm.gni") 9 import("//build/config/arm.gni")
10 import("//build/config/features.gni") 10 import("//build/config/features.gni")
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111 # http://www.openh264.org, https://www.ffmpeg.org/ 111 # http://www.openh264.org, https://www.ffmpeg.org/
112 rtc_use_h264 = proprietary_codecs && !is_android && !is_ios 112 rtc_use_h264 = proprietary_codecs && !is_android && !is_ios
113 113
114 # Determines whether QUIC code will be built. 114 # Determines whether QUIC code will be built.
115 rtc_use_quic = false 115 rtc_use_quic = false
116 116
117 # By default, use normal platform audio support or dummy audio, but don't 117 # By default, use normal platform audio support or dummy audio, but don't
118 # use file-based audio playout and record. 118 # use file-based audio playout and record.
119 rtc_use_dummy_audio_file_devices = false 119 rtc_use_dummy_audio_file_devices = false
120 120
121 # When set to true, test targets will declare the files needed to run memcheck
122 # as data dependencies. This is to enable memcheck execution on swarming bots.
123 rtc_use_memcheck = false
124
121 # FFmpeg must be initialized for |H264DecoderImpl| to work. This can be done 125 # FFmpeg must be initialized for |H264DecoderImpl| to work. This can be done
122 # by WebRTC during |H264DecoderImpl::InitDecode| or externally. FFmpeg must 126 # by WebRTC during |H264DecoderImpl::InitDecode| or externally. FFmpeg must
123 # only be initialized once. Projects that initialize FFmpeg externally, such 127 # only be initialized once. Projects that initialize FFmpeg externally, such
124 # as Chromium, must turn this flag off so that WebRTC does not also 128 # as Chromium, must turn this flag off so that WebRTC does not also
125 # initialize. 129 # initialize.
126 rtc_initialize_ffmpeg = !build_with_chromium 130 rtc_initialize_ffmpeg = !build_with_chromium
127 131
128 # Build sources requiring GTK. NOTICE: This is not present in Chrome OS 132 # Build sources requiring GTK. NOTICE: This is not present in Chrome OS
129 # build environments, even if available for Chromium builds. 133 # build environments, even if available for Chromium builds.
130 rtc_use_gtk = !build_with_chromium 134 rtc_use_gtk = !build_with_chromium
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306 ]) 310 ])
307 configs += invoker.configs 311 configs += invoker.configs
308 configs -= rtc_remove_configs 312 configs -= rtc_remove_configs
309 configs -= invoker.suppressed_configs 313 configs -= invoker.suppressed_configs
310 public_configs = [ rtc_common_inherited_config ] 314 public_configs = [ rtc_common_inherited_config ]
311 if (defined(invoker.public_configs)) { 315 if (defined(invoker.public_configs)) {
312 public_configs += invoker.public_configs 316 public_configs += invoker.public_configs
313 } 317 }
314 } 318 }
315 } 319 }
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