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Issue 2022833002: Migrate libjingle to WebRTC build targets. (Closed) Base URL: http://chromium.googlesource.com/chromium/src.git@master
Patch Set: Fixed configs Created 4 years, 6 months ago
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1 # Copyright 2014 The Chromium Authors. All rights reserved. 1 # Copyright 2014 The Chromium Authors. All rights reserved.
2 # Use of this source code is governed by a BSD-style license that can be 2 # Use of this source code is governed by a BSD-style license that can be
3 # found in the LICENSE file. 3 # found in the LICENSE file.
4 4
5 import("//build/config/features.gni") 5 import("//build/config/features.gni")
6 import("//build/config/ui.gni") 6 import("//build/config/ui.gni")
7 import("//content/renderer/renderer.gni") 7 import("//content/renderer/renderer.gni")
8 import("//media/media_options.gni") 8 import("//media/media_options.gni")
9 import("//third_party/webrtc/build/webrtc.gni") 9 import("//third_party/webrtc/build/webrtc.gni")
10 10
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160 160
161 if (enable_webrtc) { 161 if (enable_webrtc) {
162 sources += rebase_path( 162 sources += rebase_path(
163 content_renderer_gypi_values.private_renderer_webrtc_sources, 163 content_renderer_gypi_values.private_renderer_webrtc_sources,
164 ".", 164 ".",
165 "//content") 165 "//content")
166 166
167 deps += [ 167 deps += [
168 "//crypto", 168 "//crypto",
169 "//third_party/libjingle:libjingle_webrtc", 169 "//third_party/libjingle:libjingle_webrtc",
170 "//third_party/libjingle:libstunprober",
171 "//third_party/libvpx", 170 "//third_party/libvpx",
171 "//third_party/webrtc/api:libjingle_peerconnection",
172 "//third_party/webrtc/base:rtc_base", 172 "//third_party/webrtc/base:rtc_base",
173 "//third_party/webrtc/common_video", 173 "//third_party/webrtc/common_video",
174 "//third_party/webrtc/media:rtc_media",
174 "//third_party/webrtc/modules/audio_device", 175 "//third_party/webrtc/modules/audio_device",
175 "//third_party/webrtc/modules/audio_processing", 176 "//third_party/webrtc/modules/audio_processing",
176 "//third_party/webrtc/modules/video_coding:webrtc_h264", 177 "//third_party/webrtc/modules/video_coding:webrtc_h264",
178 "//third_party/webrtc/p2p:libstunprober",
179 "//third_party/webrtc/p2p:rtc_p2p",
180 "//third_party/webrtc/pc:rtc_pc",
177 "//third_party/webrtc/system_wrappers", 181 "//third_party/webrtc/system_wrappers",
178 ] 182 ]
179 if (rtc_use_h264) { 183 if (rtc_use_h264) {
180 deps += [ "//third_party/openh264:encoder" ] 184 deps += [ "//third_party/openh264:encoder" ]
181 } 185 }
182 } else { 186 } else {
183 sources += [ 187 sources += [
184 "media/webrtc_logging.h", 188 "media/webrtc_logging.h",
185 "media/webrtc_logging_noop.cc", 189 "media/webrtc_logging_noop.cc",
186 ] 190 ]
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261 # For the defines in mojo_media_config. 265 # For the defines in mojo_media_config.
262 public_configs = [ "//media/mojo/services:mojo_media_config" ] 266 public_configs = [ "//media/mojo/services:mojo_media_config" ]
263 } 267 }
264 268
265 if (!is_component_build) { 269 if (!is_component_build) {
266 public_deps = [ 270 public_deps = [
267 ":renderer", 271 ":renderer",
268 ] 272 ]
269 } 273 }
270 } 274 }
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