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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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