BUILD.gn (3890B)
1 # Copyright(c) 2020 The WebRTC project authors.All Rights Reserved. 2 # 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 5 # tree.An additional intellectual property rights grant can be found 6 # in the file PATENTS.All contributing project authors may 7 # be found in the AUTHORS file in the root of the source tree. 8 9 import("../../webrtc.gni") 10 11 rtc_library("voip_core") { 12 sources = [ 13 "voip_core.cc", 14 "voip_core.h", 15 ] 16 deps = [ 17 ":audio_channel", 18 "..:audio", 19 "../../api:array_view", 20 "../../api:make_ref_counted", 21 "../../api:scoped_refptr", 22 "../../api:transport_api", 23 "../../api/audio:audio_device", 24 "../../api/audio:audio_mixer_api", 25 "../../api/audio:audio_processing", 26 "../../api/audio_codecs:audio_codecs_api", 27 "../../api/environment", 28 "../../api/task_queue", 29 "../../api/voip:voip_api", 30 "../../call:audio_sender_interface", 31 "../../modules/audio_mixer:audio_mixer_impl", 32 "../../rtc_base:criticalsection", 33 "../../rtc_base:logging", 34 "../../rtc_base:macromagic", 35 "../../rtc_base:random", 36 "../../rtc_base:timeutils", 37 "../../rtc_base/synchronization:mutex", 38 ] 39 } 40 41 rtc_library("audio_channel") { 42 sources = [ 43 "audio_channel.cc", 44 "audio_channel.h", 45 ] 46 deps = [ 47 ":audio_egress", 48 ":audio_ingress", 49 "../../api:array_view", 50 "../../api:ref_count", 51 "../../api:rtp_headers", 52 "../../api:scoped_refptr", 53 "../../api:transport_api", 54 "../../api/audio:audio_mixer_api", 55 "../../api/audio_codecs:audio_codecs_api", 56 "../../api/environment", 57 "../../api/task_queue", 58 "../../api/voip:voip_api", 59 "../../call:audio_sender_interface", 60 "../../modules/audio_coding:audio_coding_module_typedefs", 61 "../../modules/rtp_rtcp", 62 "../../modules/rtp_rtcp:rtp_rtcp_format", 63 "../../rtc_base:checks", 64 "../../rtc_base:criticalsection", 65 "../../rtc_base:logging", 66 "../../rtc_base:refcount", 67 ] 68 } 69 70 rtc_library("audio_ingress") { 71 sources = [ 72 "audio_ingress.cc", 73 "audio_ingress.h", 74 ] 75 deps = [ 76 "..:audio", 77 "../../api:array_view", 78 "../../api:rtp_headers", 79 "../../api:scoped_refptr", 80 "../../api:transport_api", 81 "../../api/audio:audio_mixer_api", 82 "../../api/audio_codecs:audio_codecs_api", 83 "../../api/environment", 84 "../../api/neteq:default_neteq_factory", 85 "../../api/neteq:neteq_api", 86 "../../api/units:time_delta", 87 "../../api/voip:voip_api", 88 "../../modules/audio_coding", 89 "../../modules/audio_coding:audio_coding_module_typedefs", 90 "../../modules/rtp_rtcp", 91 "../../modules/rtp_rtcp:rtp_rtcp_format", 92 "../../rtc_base:checks", 93 "../../rtc_base:criticalsection", 94 "../../rtc_base:logging", 95 "../../rtc_base:macromagic", 96 "../../rtc_base:rtc_numerics", 97 "../../rtc_base:safe_conversions", 98 "../../rtc_base:safe_minmax", 99 "../../rtc_base:timeutils", 100 "../../rtc_base/synchronization:mutex", 101 "../utility:audio_frame_operations", 102 ] 103 } 104 105 rtc_library("audio_egress") { 106 sources = [ 107 "audio_egress.cc", 108 "audio_egress.h", 109 ] 110 deps = [ 111 "..:audio", 112 "../../api:array_view", 113 "../../api:sequence_checker", 114 "../../api/audio:audio_frame_api", 115 "../../api/audio_codecs:audio_codecs_api", 116 "../../api/environment", 117 "../../api/task_queue", 118 "../../call:audio_sender_interface", 119 "../../modules/audio_coding", 120 "../../modules/audio_coding:audio_coding_module_typedefs", 121 "../../modules/rtp_rtcp", 122 "../../modules/rtp_rtcp:rtp_rtcp_format", 123 "../../rtc_base:checks", 124 "../../rtc_base:logging", 125 "../../rtc_base:macromagic", 126 "../../rtc_base:safe_conversions", 127 "../../rtc_base:timeutils", 128 "../../rtc_base/synchronization:mutex", 129 "../../rtc_base/system:no_unique_address", 130 "../utility:audio_frame_operations", 131 ] 132 }