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

Issue 2943693003: Create RtcpDemuxer (Closed)
Patch Set: . Created 3 years, 6 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
(...skipping 19 matching lines...) Expand all
30 "../api/audio_codecs:audio_codecs_api", 30 "../api/audio_codecs:audio_codecs_api",
31 "../base:rtc_base", 31 "../base:rtc_base",
32 "../base:rtc_base_approved", 32 "../base:rtc_base_approved",
33 ] 33 ]
34 } 34 }
35 35
36 # TODO(nisse): These RTP targets should be moved elsewhere 36 # TODO(nisse): These RTP targets should be moved elsewhere
37 # when interfaces have stabilized. 37 # when interfaces have stabilized.
38 rtc_source_set("rtp_interfaces") { 38 rtc_source_set("rtp_interfaces") {
39 sources = [ 39 sources = [
40 "rtcp_packet_sink_interface.h",
40 "rtp_packet_sink_interface.h", 41 "rtp_packet_sink_interface.h",
41 "rtp_transport_controller_send_interface.h", 42 "rtp_transport_controller_send_interface.h",
42 ] 43 ]
43 } 44 }
44 45
45 rtc_source_set("rtp_receiver") { 46 rtc_source_set("rtp_receiver") {
46 sources = [ 47 sources = [
48 "rsid_resolution_observer.h",
49 "rtcp_demuxer.cc",
50 "rtcp_demuxer.h",
47 "rtp_demuxer.cc", 51 "rtp_demuxer.cc",
48 "rtp_demuxer.h", 52 "rtp_demuxer.h",
53 "rtp_rtcp_demuxer_helper.cc",
54 "rtp_rtcp_demuxer_helper.h",
49 "rtx_receive_stream.cc", 55 "rtx_receive_stream.cc",
50 "rtx_receive_stream.h", 56 "rtx_receive_stream.h",
51 ] 57 ]
52 deps = [ 58 deps = [
53 ":rtp_interfaces", 59 ":rtp_interfaces",
54 "../base:rtc_base_approved", 60 "../base:rtc_base_approved",
55 "../modules/rtp_rtcp", 61 "../modules/rtp_rtcp",
56 ] 62 ]
57 } 63 }
58 64
(...skipping 58 matching lines...) Expand 10 before | Expand all | Expand 10 after
117 # gets additional generated targets which would require many lines here to 123 # gets additional generated targets which would require many lines here to
118 # cover (which would be confusing to read and hard to maintain). 124 # cover (which would be confusing to read and hard to maintain).
119 if (!is_android && !is_ios) { 125 if (!is_android && !is_ios) {
120 visibility = [ "//webrtc:video_engine_tests" ] 126 visibility = [ "//webrtc:video_engine_tests" ]
121 } 127 }
122 sources = [ 128 sources = [
123 "bitrate_allocator_unittest.cc", 129 "bitrate_allocator_unittest.cc",
124 "bitrate_estimator_tests.cc", 130 "bitrate_estimator_tests.cc",
125 "call_unittest.cc", 131 "call_unittest.cc",
126 "flexfec_receive_stream_unittest.cc", 132 "flexfec_receive_stream_unittest.cc",
133 "rtcp_demuxer_unittest.cc",
127 "rtp_demuxer_unittest.cc", 134 "rtp_demuxer_unittest.cc",
135 "rtp_rtcp_demuxer_helper_unittest.cc",
128 "rtx_receive_stream_unittest.cc", 136 "rtx_receive_stream_unittest.cc",
129 ] 137 ]
130 deps = [ 138 deps = [
131 ":call", 139 ":call",
132 ":rtp_interfaces", 140 ":rtp_interfaces",
133 ":rtp_receiver", 141 ":rtp_receiver",
134 ":rtp_sender", 142 ":rtp_sender",
135 "../api:mock_audio_mixer", 143 "../api:mock_audio_mixer",
136 "../base:rtc_base_approved", 144 "../base:rtc_base_approved",
137 "../logging:rtc_event_log_api", 145 "../logging:rtc_event_log_api",
(...skipping 53 matching lines...) Expand 10 before | Expand all | Expand 10 after
191 "//testing/gtest", 199 "//testing/gtest",
192 "//webrtc/test:field_trial", 200 "//webrtc/test:field_trial",
193 "//webrtc/test:test_common", 201 "//webrtc/test:test_common",
194 ] 202 ]
195 if (!build_with_chromium && is_clang) { 203 if (!build_with_chromium && is_clang) {
196 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). 204 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
197 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] 205 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
198 } 206 }
199 } 207 }
200 } 208 }
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