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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 { | 9 { |
| 10 'includes': [ '../build/common.gypi', ], | 10 'includes': [ '../build/common.gypi', ], |
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| 92 }, | 92 }, |
| 93 ] | 93 ] |
| 94 }], | 94 }], |
| 95 ], # conditions | 95 ], # conditions |
| 96 'targets': [ | 96 'targets': [ |
| 97 { | 97 { |
| 98 'target_name': 'call_api', | 98 'target_name': 'call_api', |
| 99 'type': 'static_library', | 99 'type': 'static_library', |
| 100 'dependencies': [ | 100 'dependencies': [ |
| 101 # TODO(kjellander): Add remaining dependencies when webrtc:4243 is done. | 101 # TODO(kjellander): Add remaining dependencies when webrtc:4243 is done. |
| 102 ':transport_api', |
| 102 '<(webrtc_root)/base/base.gyp:rtc_base_approved', | 103 '<(webrtc_root)/base/base.gyp:rtc_base_approved', |
| 103 '<(webrtc_root)/common.gyp:webrtc_common', | 104 '<(webrtc_root)/common.gyp:webrtc_common', |
| 104 '<(webrtc_root)/modules/modules.gyp:audio_encoder_interface', | 105 '<(webrtc_root)/modules/modules.gyp:audio_encoder_interface', |
| 105 ], | 106 ], |
| 106 'sources': [ | 107 'sources': [ |
| 107 'call/audio_receive_stream.h', | 108 'call/audio_receive_stream.h', |
| 108 'call/audio_send_stream.h', | 109 'call/audio_send_stream.h', |
| 109 'call/audio_sink.h', | 110 'call/audio_sink.h', |
| 110 'call/audio_state.h', | 111 'call/audio_state.h', |
| 111 ], | 112 ], |
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| 224 'type': 'static_library', | 225 'type': 'static_library', |
| 225 'dependencies': [ | 226 'dependencies': [ |
| 226 '<(webrtc_root)/base/base.gyp:rtc_base_approved', | 227 '<(webrtc_root)/base/base.gyp:rtc_base_approved', |
| 227 ], | 228 ], |
| 228 'sources': [ | 229 'sources': [ |
| 229 'stats/rtcstats.h', | 230 'stats/rtcstats.h', |
| 230 'stats/rtcstats_objects.h', | 231 'stats/rtcstats_objects.h', |
| 231 'stats/rtcstatsreport.h', | 232 'stats/rtcstatsreport.h', |
| 232 ], | 233 ], |
| 233 }, # target rtc_stats_api | 234 }, # target rtc_stats_api |
| 235 { |
| 236 # GN version: webrtc/api:transport_api |
| 237 'target_name': 'transport_api', |
| 238 'type': 'static_library', |
| 239 'sources': [ |
| 240 'call/transport.h', |
| 241 ], |
| 242 } |
| 234 ], # targets | 243 ], # targets |
| 235 } | 244 } |
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