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

Issue 2809613002: Revert of Implemented the GetSources() in native code. (Closed)
Patch Set: Created 3 years, 8 months ago
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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
11 rtc_source_set("call_interfaces") { 11 rtc_source_set("call_interfaces") {
12 sources = [ 12 sources = [
13 "audio_receive_stream.h", 13 "audio_receive_stream.h",
14 "audio_send_stream.cc", 14 "audio_send_stream.cc",
15 "audio_send_stream.h", 15 "audio_send_stream.h",
16 "audio_state.h", 16 "audio_state.h",
17 "call.h", 17 "call.h",
18 "flexfec_receive_stream.h", 18 "flexfec_receive_stream.h",
19 "rtp_transport_controller_send.h", 19 "rtp_transport_controller_send.h",
20 "syncable.cc", 20 "syncable.cc",
21 "syncable.h", 21 "syncable.h",
22 ] 22 ]
23 deps = [ 23 deps = [
24 "..:webrtc_common", 24 "..:webrtc_common",
25 "../api:audio_mixer_api", 25 "../api:audio_mixer_api",
26 "../api:libjingle_peerconnection_api",
27 "../api:transport_api", 26 "../api:transport_api",
28 "../api/audio_codecs:audio_codecs_api", 27 "../api/audio_codecs:audio_codecs_api",
29 "../base:rtc_base", 28 "../base:rtc_base",
30 "../base:rtc_base_approved", 29 "../base:rtc_base_approved",
31 "../modules/audio_coding:audio_encoder_interface", 30 "../modules/audio_coding:audio_encoder_interface",
32 ] 31 ]
33 } 32 }
34 33
35 rtc_static_library("call") { 34 rtc_static_library("call") {
36 sources = [ 35 sources = [
(...skipping 89 matching lines...) Expand 10 before | Expand all | Expand 10 after
126 "//testing/gtest", 125 "//testing/gtest",
127 "//webrtc/test:field_trial", 126 "//webrtc/test:field_trial",
128 "//webrtc/test:test_common", 127 "//webrtc/test:test_common",
129 ] 128 ]
130 if (!build_with_chromium && is_clang) { 129 if (!build_with_chromium && is_clang) {
131 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). 130 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
132 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] 131 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
133 } 132 }
134 } 133 }
135 } 134 }
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