Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(66)

Side by Side Diff: webrtc/call/BUILD.gn

Issue 3008553002: rtc_task_queue should not expose rtc_base_approved API (Closed)
Patch Set: Created 3 years, 3 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « no previous file | webrtc/modules/utility/BUILD.gn » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 # Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 1 # Copyright (c) 2015 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("../webrtc.gni") 9 import("../webrtc.gni")
10 10
(...skipping 99 matching lines...) Expand 10 before | Expand all | Expand 10 after
110 "..:webrtc_common", 110 "..:webrtc_common",
111 "../api:transport_api", 111 "../api:transport_api",
112 "../audio", 112 "../audio",
113 "../logging:rtc_event_log_api", 113 "../logging:rtc_event_log_api",
114 "../logging:rtc_event_log_impl", 114 "../logging:rtc_event_log_impl",
115 "../modules/bitrate_controller", 115 "../modules/bitrate_controller",
116 "../modules/congestion_controller", 116 "../modules/congestion_controller",
117 "../modules/pacing", 117 "../modules/pacing",
118 "../modules/rtp_rtcp", 118 "../modules/rtp_rtcp",
119 "../modules/utility", 119 "../modules/utility",
120 "../rtc_base:rtc_base_approved",
120 "../rtc_base:rtc_task_queue", 121 "../rtc_base:rtc_task_queue",
121 "../rtc_base:sequenced_task_checker", 122 "../rtc_base:sequenced_task_checker",
122 "../system_wrappers", 123 "../system_wrappers",
123 "../video", 124 "../video",
124 ] 125 ]
125 } 126 }
126 127
127 rtc_source_set("video_stream_api") { 128 rtc_source_set("video_stream_api") {
128 sources = [ 129 sources = [
129 "video_receive_stream.cc", 130 "video_receive_stream.cc",
(...skipping 110 matching lines...) Expand 10 before | Expand all | Expand 10 after
240 sources = [ 241 sources = [
241 "test/mock_rtp_packet_sink_interface.h", 242 "test/mock_rtp_packet_sink_interface.h",
242 ] 243 ]
243 deps = [ 244 deps = [
244 ":rtp_interfaces", 245 ":rtp_interfaces",
245 "../test:test_support", 246 "../test:test_support",
246 "//testing/gmock", 247 "//testing/gmock",
247 ] 248 ]
248 } 249 }
249 } 250 }
OLDNEW
« no previous file with comments | « no previous file | webrtc/modules/utility/BUILD.gn » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698