Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(3)

Side by Side Diff: webrtc/api/api.gyp

Issue 2059703002: Move webrtc/audio_*.h to webrtc/api/call (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fixed audio_sink.h Created 4 years, 6 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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', ],
(...skipping 105 matching lines...) Expand 10 before | Expand all | Expand 10 after
116 116
117 ], 117 ],
118 }, 118 },
119 'includes': ['../../build/java.gypi'], 119 'includes': ['../../build/java.gypi'],
120 }, # libjingle_peerconnection_java 120 }, # libjingle_peerconnection_java
121 ] 121 ]
122 }], 122 }],
123 ], # conditions 123 ], # conditions
124 'targets': [ 124 'targets': [
125 { 125 {
126 'target_name': 'audio_api',
127 'type': 'static_library',
128 'dependencies': [
129 # TODO(kjellander): Add remaining dependencies when webrtc:4243 is done.
130 '<(webrtc_root)/base/base.gyp:rtc_base_approved',
131 '<(webrtc_root)/common.gyp:webrtc_common',
132 '<(webrtc_root)/modules/modules.gyp:audio_encoder_interface',
133 ],
134 'sources': [
135 'audio/audio_receive_stream.h',
136 'audio/audio_send_stream.h',
137 'audio/audio_sink.h',
138 'audio/audio_state.h',
139 ],
140 },
141 {
126 'target_name': 'libjingle_peerconnection', 142 'target_name': 'libjingle_peerconnection',
127 'type': 'static_library', 143 'type': 'static_library',
128 'dependencies': [ 144 'dependencies': [
145 ':audio_api',
129 '<(webrtc_root)/media/media.gyp:rtc_media', 146 '<(webrtc_root)/media/media.gyp:rtc_media',
130 '<(webrtc_root)/pc/pc.gyp:rtc_pc', 147 '<(webrtc_root)/pc/pc.gyp:rtc_pc',
131 ], 148 ],
132 'sources': [ 149 'sources': [
133 'audiotrack.cc', 150 'audiotrack.cc',
134 'audiotrack.h', 151 'audiotrack.h',
135 'datachannel.cc', 152 'datachannel.cc',
136 'datachannel.h', 153 'datachannel.h',
137 'datachannelinterface.h', 154 'datachannelinterface.h',
138 'dtlsidentitystore.h', 155 'dtlsidentitystore.h',
(...skipping 105 matching lines...) Expand 10 before | Expand all | Expand 10 after
244 'quicdatatransport.h', 261 'quicdatatransport.h',
245 ], 262 ],
246 'export_dependent_settings': [ 263 'export_dependent_settings': [
247 '<(DEPTH)/third_party/libquic/libquic.gyp:libquic', 264 '<(DEPTH)/third_party/libquic/libquic.gyp:libquic',
248 ], 265 ],
249 }], 266 }],
250 ], 267 ],
251 }, # target libjingle_peerconnection 268 }, # target libjingle_peerconnection
252 ], # targets 269 ], # targets
253 } 270 }
OLDNEW
« webrtc/api/BUILD.gn ('K') | « webrtc/api/BUILD.gn ('k') | webrtc/api/audio/DEPS » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698