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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_demuxer.h", | 19 "rtp_demuxer.h", |
| 20 "rtp_transport_controller_receive_interface.h", |
20 "rtp_transport_controller_send_interface.h", | 21 "rtp_transport_controller_send_interface.h", |
21 "syncable.cc", | 22 "syncable.cc", |
22 "syncable.h", | 23 "syncable.h", |
23 ] | 24 ] |
24 deps = [ | 25 deps = [ |
25 "..:video_stream_api", | 26 "..:video_stream_api", |
26 "..:webrtc_common", | 27 "..:webrtc_common", |
27 "../api:audio_mixer_api", | 28 "../api:audio_mixer_api", |
28 "../api:libjingle_peerconnection_api", | 29 "../api:libjingle_peerconnection_api", |
29 "../api:transport_api", | 30 "../api:transport_api", |
30 "../api/audio_codecs:audio_codecs_api", | 31 "../api/audio_codecs:audio_codecs_api", |
31 "../base:rtc_base", | 32 "../base:rtc_base", |
32 "../base:rtc_base_approved", | 33 "../base:rtc_base_approved", |
33 ] | 34 ] |
34 } | 35 } |
35 | 36 |
36 rtc_static_library("call") { | 37 rtc_static_library("call") { |
37 sources = [ | 38 sources = [ |
38 "bitrate_allocator.cc", | 39 "bitrate_allocator.cc", |
39 "call.cc", | 40 "call.cc", |
40 "flexfec_receive_stream_impl.cc", | 41 "flexfec_receive_stream_impl.cc", |
41 "flexfec_receive_stream_impl.h", | 42 "flexfec_receive_stream_impl.h", |
42 "rtp_demuxer.cc", | 43 "rtp_demuxer.cc", |
| 44 "rtp_transport_controller_receive.cc", |
| 45 "rtp_transport_controller_receive.h", |
43 "rtp_transport_controller_send.cc", | 46 "rtp_transport_controller_send.cc", |
44 "rtp_transport_controller_send.h", | 47 "rtp_transport_controller_send.h", |
45 ] | 48 ] |
46 | 49 |
47 if (!build_with_chromium && is_clang) { | 50 if (!build_with_chromium && is_clang) { |
48 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 51 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
49 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 52 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
50 } | 53 } |
51 | 54 |
52 public_deps = [ | 55 public_deps = [ |
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148 "//testing/gtest", | 151 "//testing/gtest", |
149 "//webrtc/test:field_trial", | 152 "//webrtc/test:field_trial", |
150 "//webrtc/test:test_common", | 153 "//webrtc/test:test_common", |
151 ] | 154 ] |
152 if (!build_with_chromium && is_clang) { | 155 if (!build_with_chromium && is_clang) { |
153 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 156 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
154 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 157 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
155 } | 158 } |
156 } | 159 } |
157 } | 160 } |
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