| OLD | NEW |
|---|
| (Empty) | |
| 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 'SRTP_RELATIVE_PATH', |
| 14 'HAVE_SCTP', |
| 15 'HAVE_SRTP', |
| 16 ], |
| 17 }, |
| 18 'targets': [ |
| 19 { |
| 20 'target_name': 'rtc_pc', |
| 21 'type': 'static_library', |
| 22 'dependencies': [ |
| 23 '<(webrtc_root)/base/base.gyp:rtc_base', |
| 24 '<(webrtc_root)/media/media.gyp:rtc_media', |
| 25 ], |
| 26 'conditions': [ |
| 27 ['build_libsrtp==1', { |
| 28 'dependencies': [ |
| 29 '<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp', |
| 30 ], |
| 31 }], |
| 32 ], |
| 33 'defines': [ |
| 34 '<@(rtc_pc_defines)', |
| 35 ], |
| 36 # TODO(kjellander): Make the code compile without disabling these flags. |
| 37 # See https://bugs.chromium.org/p/webrtc/issues/detail?id=3307 |
| 38 'cflags_cc!': [ |
| 39 '-Wnon-virtual-dtor', |
| 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 'mediasink.h', |
| 68 'rtcpmuxfilter.cc', |
| 69 'rtcpmuxfilter.h', |
| 70 'srtpfilter.cc', |
| 71 'srtpfilter.h', |
| 72 'voicechannel.h', |
| 73 ], |
| 74 }, # target rtc_pc |
| 75 ], |
| 76 } |
| OLD | NEW |
|---|
Chromium Code Reviews
Issue 1691463002:
Move talk/session/media -> webrtc/pc (Closed)
Base URL: https://chromium.googlesource.com/external/webrtc.git@master