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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 "rtp_transport_controller_send.h", | 20 "rtp_transport_controller_send.h", |
20 "syncable.cc", | 21 "syncable.cc", |
21 "syncable.h", | 22 "syncable.h", |
22 ] | 23 ] |
23 deps = [ | 24 deps = [ |
24 "..:webrtc_common", | 25 "..:webrtc_common", |
25 "../api:audio_mixer_api", | 26 "../api:audio_mixer_api", |
26 "../api:transport_api", | 27 "../api:transport_api", |
27 "../api/audio_codecs:audio_codecs_api", | 28 "../api/audio_codecs:audio_codecs_api", |
28 "../base:rtc_base", | 29 "../base:rtc_base", |
29 "../base:rtc_base_approved", | 30 "../base:rtc_base_approved", |
30 "../modules/audio_coding:audio_encoder_interface", | 31 "../modules/audio_coding:audio_encoder_interface", |
31 ] | 32 ] |
32 } | 33 } |
33 | 34 |
34 rtc_static_library("call") { | 35 rtc_static_library("call") { |
35 sources = [ | 36 sources = [ |
36 "bitrate_allocator.cc", | 37 "bitrate_allocator.cc", |
37 "call.cc", | 38 "call.cc", |
38 "flexfec_receive_stream_impl.cc", | 39 "flexfec_receive_stream_impl.cc", |
39 "flexfec_receive_stream_impl.h", | 40 "flexfec_receive_stream_impl.h", |
| 41 "rtp_transport_controller_receive.cc", |
40 ] | 42 ] |
41 | 43 |
42 if (!build_with_chromium && is_clang) { | 44 if (!build_with_chromium && is_clang) { |
43 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 45 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
44 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 46 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
45 } | 47 } |
46 | 48 |
47 public_deps = [ | 49 public_deps = [ |
48 ":call_interfaces", | 50 ":call_interfaces", |
49 "../api:call_api", | 51 "../api:call_api", |
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125 "//testing/gtest", | 127 "//testing/gtest", |
126 "//webrtc/test:field_trial", | 128 "//webrtc/test:field_trial", |
127 "//webrtc/test:test_common", | 129 "//webrtc/test:test_common", |
128 ] | 130 ] |
129 if (!build_with_chromium && is_clang) { | 131 if (!build_with_chromium && is_clang) { |
130 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 132 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
131 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 133 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
132 } | 134 } |
133 } | 135 } |
134 } | 136 } |
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