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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/mips.gni") | 10 import("//build/config/mips.gni") |
| 11 import("//build_overrides/webrtc.gni") |
| 11 | 12 |
| 12 declare_args() { | 13 declare_args() { |
| 13 # Assume Chromium build for now, since that's the priority case for getting GN | |
| 14 # up and running with WebRTC. | |
| 15 build_with_chromium = true | |
| 16 build_with_libjingle = true | 14 build_with_libjingle = true |
| 17 | 15 |
| 18 # Disable this to avoid building the Opus audio codec. | 16 # Disable this to avoid building the Opus audio codec. |
| 19 rtc_include_opus = true | 17 rtc_include_opus = true |
| 20 | 18 |
| 21 # Used to specify an external Jsoncpp include path when not compiling the | 19 # Used to specify an external Jsoncpp include path when not compiling the |
| 22 # library that comes with WebRTC (i.e. rtc_build_json == 0). | 20 # library that comes with WebRTC (i.e. rtc_build_json == 0). |
| 23 rtc_jsoncpp_root = "//third_party/jsoncpp/source/include" | 21 rtc_jsoncpp_root = "//third_party/jsoncpp/source/include" |
| 24 | 22 |
| 25 # Used to specify an external OpenSSL include path when not compiling the | 23 # Used to specify an external OpenSSL include path when not compiling the |
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| 54 # Enable to use the Mozilla internal settings. | 52 # Enable to use the Mozilla internal settings. |
| 55 build_with_mozilla = false | 53 build_with_mozilla = false |
| 56 | 54 |
| 57 rtc_enable_android_opensl = false | 55 rtc_enable_android_opensl = false |
| 58 | 56 |
| 59 # Link-Time Optimizations. | 57 # Link-Time Optimizations. |
| 60 # Executes code generation at link-time instead of compile-time. | 58 # Executes code generation at link-time instead of compile-time. |
| 61 # https://gcc.gnu.org/wiki/LinkTimeOptimization | 59 # https://gcc.gnu.org/wiki/LinkTimeOptimization |
| 62 rtc_use_lto = false | 60 rtc_use_lto = false |
| 63 | 61 |
| 64 if (build_with_chromium) { | |
| 65 # Exclude pulse audio on Chromium since its prerequisites don't require | |
| 66 # pulse audio. | |
| 67 rtc_include_pulse_audio = false | |
| 68 | |
| 69 # Exclude internal ADM since Chromium uses its own IO handling. | |
| 70 rtc_include_internal_audio_device = false | |
| 71 } else { | |
| 72 # Settings for the standalone (not-in-Chromium) build. | |
| 73 | |
| 74 # TODO(andrew): For now, disable the Chrome plugins, which causes a | |
| 75 # flood of chromium-style warnings. Investigate enabling them: | |
| 76 # http://code.google.com/p/webrtc/issues/detail?id=163 | |
| 77 clang_use_chrome_plugins = false | |
| 78 | |
| 79 rtc_include_pulse_audio = true | |
| 80 rtc_include_internal_audio_device = true | |
| 81 } | |
| 82 | |
| 83 if (build_with_libjingle) { | 62 if (build_with_libjingle) { |
| 84 rtc_include_tests = false | 63 rtc_include_tests = false |
| 85 rtc_restrict_logging = true | 64 rtc_restrict_logging = true |
| 86 } else { | 65 } else { |
| 87 rtc_include_tests = true | 66 rtc_include_tests = true |
| 88 rtc_restrict_logging = false | 67 rtc_restrict_logging = false |
| 89 } | 68 } |
| 90 | 69 |
| 91 if (is_ios) { | 70 if (is_ios) { |
| 92 rtc_build_libjpeg = false | 71 rtc_build_libjpeg = false |
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| 113 # Enable this to use HW H.264 encoder/decoder on iOS PeerConnections. | 92 # Enable this to use HW H.264 encoder/decoder on iOS PeerConnections. |
| 114 # Enabling this may break interop with Android clients that support H264. | 93 # Enabling this may break interop with Android clients that support H264. |
| 115 rtc_use_objc_h264 = false | 94 rtc_use_objc_h264 = false |
| 116 } | 95 } |
| 117 | 96 |
| 118 # Make it possible to provide custom locations for some libraries (move these | 97 # Make it possible to provide custom locations for some libraries (move these |
| 119 # up into declare_args should we need to actually use them for the GN build). | 98 # up into declare_args should we need to actually use them for the GN build). |
| 120 rtc_libvpx_dir = "//third_party/libvpx_new" | 99 rtc_libvpx_dir = "//third_party/libvpx_new" |
| 121 rtc_libyuv_dir = "//third_party/libyuv" | 100 rtc_libyuv_dir = "//third_party/libyuv" |
| 122 rtc_opus_dir = "//third_party/opus" | 101 rtc_opus_dir = "//third_party/opus" |
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