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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 | 
| (...skipping 28 matching lines...) Expand all  Loading... | 
| 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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