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1 # Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. | 1 # Copyright (c) 2011 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': [ | 10 'includes': [ |
11 '../build/common.gypi', | 11 '../build/common.gypi', |
12 ], | 12 ], |
13 'targets': [ | 13 'targets': [ |
14 { | 14 { |
15 'target_name': 'voice_engine', | 15 'target_name': 'voice_engine', |
16 'type': 'static_library', | 16 'type': 'static_library', |
17 'dependencies': [ | 17 'dependencies': [ |
18 '<(webrtc_root)/common.gyp:webrtc_common', | 18 '<(webrtc_root)/common.gyp:webrtc_common', |
19 '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', | 19 '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', |
20 '<(webrtc_root)/modules/modules.gyp:audio_coding_module', | 20 '<(webrtc_root)/modules/modules.gyp:audio_coding_module', |
21 '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer', | 21 '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer', |
22 '<(webrtc_root)/modules/modules.gyp:audio_device', | 22 '<(webrtc_root)/modules/modules.gyp:audio_device', |
23 '<(webrtc_root)/modules/modules.gyp:audio_processing', | 23 '<(webrtc_root)/modules/modules.gyp:audio_processing', |
24 '<(webrtc_root)/modules/modules.gyp:bitrate_controller', | 24 '<(webrtc_root)/modules/modules.gyp:bitrate_controller', |
25 '<(webrtc_root)/modules/modules.gyp:media_file', | 25 '<(webrtc_root)/modules/modules.gyp:media_file', |
| 26 '<(webrtc_root)/modules/modules.gyp:paced_sender', |
26 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp', | 27 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp', |
27 '<(webrtc_root)/modules/modules.gyp:webrtc_utility', | 28 '<(webrtc_root)/modules/modules.gyp:webrtc_utility', |
28 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', | 29 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', |
29 '<(webrtc_root)/webrtc.gyp:rtc_event_log', | 30 '<(webrtc_root)/webrtc.gyp:rtc_event_log', |
30 ], | 31 ], |
31 'export_dependent_settings': [ | 32 'export_dependent_settings': [ |
32 '<(webrtc_root)/modules/modules.gyp:audio_coding_module', | 33 '<(webrtc_root)/modules/modules.gyp:audio_coding_module', |
33 ], | 34 ], |
34 'sources': [ | 35 'sources': [ |
35 'include/voe_audio_processing.h', | 36 'include/voe_audio_processing.h', |
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288 'sources': [ | 289 'sources': [ |
289 'voe_auto_test.isolate', | 290 'voe_auto_test.isolate', |
290 ], | 291 ], |
291 }, | 292 }, |
292 ], | 293 ], |
293 }], | 294 }], |
294 ], # conditions | 295 ], # conditions |
295 }], # include_tests==1 | 296 }], # include_tests==1 |
296 ], # conditions | 297 ], # conditions |
297 } | 298 } |
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