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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 |
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21 "syncable.h", | 21 "syncable.h", |
22 ] | 22 ] |
23 deps = [ | 23 deps = [ |
24 "..:webrtc_common", | 24 "..:webrtc_common", |
25 "../api:audio_mixer_api", | 25 "../api:audio_mixer_api", |
26 "../api:libjingle_peerconnection_api", | 26 "../api:libjingle_peerconnection_api", |
27 "../api:transport_api", | 27 "../api:transport_api", |
28 "../api/audio_codecs:audio_codecs_api", | 28 "../api/audio_codecs:audio_codecs_api", |
29 "../base:rtc_base", | 29 "../base:rtc_base", |
30 "../base:rtc_base_approved", | 30 "../base:rtc_base_approved", |
| 31 "../modules/audio_coding:audio_encoder_factory_interface", |
31 "../modules/audio_coding:audio_encoder_interface", | 32 "../modules/audio_coding:audio_encoder_interface", |
32 ] | 33 ] |
33 } | 34 } |
34 | 35 |
35 rtc_static_library("call") { | 36 rtc_static_library("call") { |
36 sources = [ | 37 sources = [ |
37 "bitrate_allocator.cc", | 38 "bitrate_allocator.cc", |
38 "call.cc", | 39 "call.cc", |
39 "flexfec_receive_stream_impl.cc", | 40 "flexfec_receive_stream_impl.cc", |
40 "flexfec_receive_stream_impl.h", | 41 "flexfec_receive_stream_impl.h", |
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107 "call_perf_tests.cc", | 108 "call_perf_tests.cc", |
108 "rampup_tests.cc", | 109 "rampup_tests.cc", |
109 "rampup_tests.h", | 110 "rampup_tests.h", |
110 ] | 111 ] |
111 deps = [ | 112 deps = [ |
112 ":call_interfaces", | 113 ":call_interfaces", |
113 "..:webrtc_common", | 114 "..:webrtc_common", |
114 "../base:rtc_base_approved", | 115 "../base:rtc_base_approved", |
115 "../logging:rtc_event_log_api", | 116 "../logging:rtc_event_log_api", |
116 "../modules/audio_coding", | 117 "../modules/audio_coding", |
| 118 "../modules/audio_coding:builtin_audio_encoder_factory", |
117 "../modules/audio_mixer:audio_mixer_impl", | 119 "../modules/audio_mixer:audio_mixer_impl", |
118 "../modules/rtp_rtcp", | 120 "../modules/rtp_rtcp", |
119 "../system_wrappers", | 121 "../system_wrappers", |
120 "../system_wrappers:metrics_default", | 122 "../system_wrappers:metrics_default", |
121 "../test:direct_transport", | 123 "../test:direct_transport", |
122 "../test:fake_audio_device", | 124 "../test:fake_audio_device", |
123 "../test:test_support", | 125 "../test:test_support", |
124 "../test:video_test_common", | 126 "../test:video_test_common", |
125 "../video", | 127 "../video", |
126 "../voice_engine", | 128 "../voice_engine", |
127 "//testing/gtest", | 129 "//testing/gtest", |
128 "//webrtc/test:field_trial", | 130 "//webrtc/test:field_trial", |
129 "//webrtc/test:test_common", | 131 "//webrtc/test:test_common", |
130 ] | 132 ] |
131 if (!build_with_chromium && is_clang) { | 133 if (!build_with_chromium && is_clang) { |
132 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 134 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
133 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 135 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
134 } | 136 } |
135 } | 137 } |
136 } | 138 } |
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