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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 ':audio_mixer_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.cc', | 109 'call/audio_send_stream.cc', |
109 'call/audio_send_stream.h', | 110 'call/audio_send_stream.h', |
110 'call/audio_sink.h', | 111 'call/audio_sink.h', |
111 'call/audio_state.h', | 112 'call/audio_state.h', |
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239 'type': 'static_library', | 240 'type': 'static_library', |
240 'dependencies': [ | 241 'dependencies': [ |
241 '<(webrtc_root)/base/base.gyp:rtc_base_approved', | 242 '<(webrtc_root)/base/base.gyp:rtc_base_approved', |
242 ], | 243 ], |
243 'sources': [ | 244 'sources': [ |
244 'audio/audio_mixer.h', | 245 'audio/audio_mixer.h', |
245 ], | 246 ], |
246 }, # target rtc_stats_api | 247 }, # target rtc_stats_api |
247 ], # targets | 248 ], # targets |
248 } | 249 } |
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