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1 # Copyright (c) 2016 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 { | |
10 'includes': ['../build/common.gypi'], | |
11 'variables': { | |
12 'rtc_pc_defines': [ | |
13 'HAVE_SCTP', | |
14 'HAVE_SRTP', | |
15 ], | |
16 }, | |
17 'targets': [ | |
18 { | |
19 'target_name': 'rtc_pc', | |
20 'type': 'static_library', | |
21 'dependencies': [ | |
22 '<(webrtc_root)/base/base.gyp:rtc_base_approved', | |
23 '<(webrtc_root)/media/media.gyp:rtc_media', | |
24 ], | |
25 'conditions': [ | |
26 ['build_with_chromium==1', { | |
27 'sources': [ | |
28 'externalhmac.h', | |
29 'externalhmac.cc', | |
30 ], | |
31 }], | |
32 ['build_libsrtp==1', { | |
33 'dependencies': [ | |
34 '<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp', | |
35 ], | |
36 }], | |
37 ], | |
38 'defines': [ | |
39 '<@(rtc_pc_defines)', | |
40 ], | |
41 'include_dirs': [ | |
42 '<(DEPTH)/testing/gtest/include', | |
43 ], | |
44 'direct_dependent_settings': { | |
45 'defines': [ | |
46 '<@(rtc_pc_defines)' | |
47 ], | |
48 'include_dirs': [ | |
49 '<(DEPTH)/testing/gtest/include', | |
50 ], | |
51 }, | |
52 'sources': [ | |
53 'audiomonitor.cc', | |
54 'audiomonitor.h', | |
55 'bundlefilter.cc', | |
56 'bundlefilter.h', | |
57 'channel.cc', | |
58 'channel.h', | |
59 'channelmanager.cc', | |
60 'channelmanager.h', | |
61 'currentspeakermonitor.cc', | |
62 'currentspeakermonitor.h', | |
63 'mediamonitor.cc', | |
64 'mediamonitor.h', | |
65 'mediasession.cc', | |
66 'mediasession.h', | |
67 'rtcpmuxfilter.cc', | |
68 'rtcpmuxfilter.h', | |
69 'srtpfilter.cc', | |
70 'srtpfilter.h', | |
71 'voicechannel.h', | |
72 ], | |
73 }, # target rtc_pc | |
74 ], # targets | |
75 } | |
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