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Issue 2397843005: Add FlexfecReceiveStream. (Closed)
Patch Set: Feedback response 1. Created 4 years, 2 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("../build/webrtc.gni") 9 import("../build/webrtc.gni")
10 if (is_android) { 10 if (is_android) {
11 import("//build/config/android/config.gni") 11 import("//build/config/android/config.gni")
12 import("//build/config/android/rules.gni") 12 import("//build/config/android/rules.gni")
13 } 13 }
14 14
15 group("api") { 15 group("api") {
16 public_deps = [ 16 public_deps = [
17 ":libjingle_peerconnection", 17 ":libjingle_peerconnection",
18 ] 18 ]
19 } 19 }
20 20
21 rtc_source_set("call_api") { 21 rtc_source_set("call_api") {
22 sources = [ 22 sources = [
23 "call/audio_receive_stream.h", 23 "call/audio_receive_stream.h",
24 "call/audio_send_stream.h", 24 "call/audio_send_stream.h",
25 "call/audio_sink.h", 25 "call/audio_sink.h",
26 "call/audio_state.h", 26 "call/audio_state.h",
27 "call/flexfec_receive_stream.h",
27 ] 28 ]
28 29
29 deps = [ 30 deps = [
30 # TODO(kjellander): Add remaining dependencies when webrtc:4243 is done. 31 # TODO(kjellander): Add remaining dependencies when webrtc:4243 is done.
31 "..:webrtc_common", 32 "..:webrtc_common",
32 "../base:rtc_base_approved", 33 "../base:rtc_base_approved",
33 "../modules/audio_coding:audio_encoder_interface", 34 "../modules/audio_coding:audio_encoder_interface",
34 ] 35 ]
35 } 36 }
36 37
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110 "webrtcsdp.cc", 111 "webrtcsdp.cc",
111 "webrtcsdp.h", 112 "webrtcsdp.h",
112 "webrtcsession.cc", 113 "webrtcsession.cc",
113 "webrtcsession.h", 114 "webrtcsession.h",
114 "webrtcsessiondescriptionfactory.cc", 115 "webrtcsessiondescriptionfactory.cc",
115 "webrtcsessiondescriptionfactory.h", 116 "webrtcsessiondescriptionfactory.h",
116 ] 117 ]
117 118
118 configs += [ ":libjingle_peerconnection_warnings_config" ] 119 configs += [ ":libjingle_peerconnection_warnings_config" ]
119 120
120 if (!build_with_chromium && is_clang) { 121 if (!build_with_chromium && is_clang) {
brandtr 2016/10/18 14:31:18 Accidental rebase!
121 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). 122 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
122 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] 123 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
123 } 124 }
124 125
125 deps = [ 126 deps = [
126 ":call_api", 127 ":call_api",
127 ":rtc_stats_api", 128 ":rtc_stats_api",
128 "../call", 129 "../call",
129 "../media", 130 "../media",
130 "../pc", 131 "../pc",
(...skipping 334 matching lines...) Expand 10 before | Expand all | Expand 10 after
465 466
466 shared_libraries = [ ":libjingle_peerconnection_so" ] 467 shared_libraries = [ ":libjingle_peerconnection_so" ]
467 } 468 }
468 469
469 android_resources("libjingle_peerconnection_android_unittest_resources") { 470 android_resources("libjingle_peerconnection_android_unittest_resources") {
470 resource_dirs = [ "androidtests/res" ] 471 resource_dirs = [ "androidtests/res" ]
471 custom_package = "org.webrtc" 472 custom_package = "org.webrtc"
472 } 473 }
473 } 474 }
474 } 475 }
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