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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_receive.h", |
19 "syncable.cc", | 20 "syncable.cc", |
20 "syncable.h", | 21 "syncable.h", |
21 ] | 22 ] |
22 deps = [ | 23 deps = [ |
23 "..:webrtc_common", | 24 "..:webrtc_common", |
24 "../api:audio_mixer_api", | 25 "../api:audio_mixer_api", |
25 "../api:transport_api", | 26 "../api:transport_api", |
26 "../api/audio_codecs:audio_codecs_api", | 27 "../api/audio_codecs:audio_codecs_api", |
27 "../base:rtc_base", | 28 "../base:rtc_base", |
28 "../base:rtc_base_approved", | 29 "../base:rtc_base_approved", |
29 "../modules/audio_coding:audio_encoder_interface", | 30 "../modules/audio_coding:audio_encoder_interface", |
30 ] | 31 ] |
31 } | 32 } |
32 | 33 |
33 rtc_static_library("call") { | 34 rtc_static_library("call") { |
34 sources = [ | 35 sources = [ |
35 "bitrate_allocator.cc", | 36 "bitrate_allocator.cc", |
36 "call.cc", | 37 "call.cc", |
37 "flexfec_receive_stream_impl.cc", | 38 "flexfec_receive_stream_impl.cc", |
38 "flexfec_receive_stream_impl.h", | 39 "flexfec_receive_stream_impl.h", |
| 40 "rtp_transport_controller_receive.cc", |
39 ] | 41 ] |
40 | 42 |
41 if (!build_with_chromium && is_clang) { | 43 if (!build_with_chromium && is_clang) { |
42 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 44 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
43 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 45 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
44 } | 46 } |
45 | 47 |
46 public_deps = [ | 48 public_deps = [ |
47 ":call_interfaces", | 49 ":call_interfaces", |
48 "../api:call_api", | 50 "../api:call_api", |
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123 "../voice_engine", | 125 "../voice_engine", |
124 "//testing/gtest", | 126 "//testing/gtest", |
125 "//webrtc/test:test_common", | 127 "//webrtc/test:test_common", |
126 ] | 128 ] |
127 if (!build_with_chromium && is_clang) { | 129 if (!build_with_chromium && is_clang) { |
128 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 130 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
129 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 131 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
130 } | 132 } |
131 } | 133 } |
132 } | 134 } |
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