Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(73)

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

Issue 2092633002: Revert of Move FilePlayer and FileRecorder to Voice Engine (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@remove0
Patch Set: 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
« no previous file with comments | « webrtc/voice_engine/transmit_mixer.h ('k') | no next file » | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
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': [
(...skipping 34 matching lines...) Expand 10 before | Expand all | Expand 10 after
45 'include/voe_network.h', 45 'include/voe_network.h',
46 'include/voe_rtp_rtcp.h', 46 'include/voe_rtp_rtcp.h',
47 'include/voe_video_sync.h', 47 'include/voe_video_sync.h',
48 'include/voe_volume_control.h', 48 'include/voe_volume_control.h',
49 'channel.cc', 49 'channel.cc',
50 'channel.h', 50 'channel.h',
51 'channel_manager.cc', 51 'channel_manager.cc',
52 'channel_manager.h', 52 'channel_manager.h',
53 'channel_proxy.cc', 53 'channel_proxy.cc',
54 'channel_proxy.h', 54 'channel_proxy.h',
55 'coder.cc',
56 'coder.h',
57 'file_player.h',
58 'file_player_impl.cc',
59 'file_player_impl.h',
60 'file_recorder.h',
61 'file_recorder_impl.cc',
62 'file_recorder_impl.h',
63 'level_indicator.cc', 55 'level_indicator.cc',
64 'level_indicator.h', 56 'level_indicator.h',
65 'monitor_module.cc', 57 'monitor_module.cc',
66 'monitor_module.h', 58 'monitor_module.h',
67 'network_predictor.cc', 59 'network_predictor.cc',
68 'network_predictor.h', 60 'network_predictor.h',
69 'output_mixer.cc', 61 'output_mixer.cc',
70 'output_mixer.h', 62 'output_mixer.h',
71 'shared_data.cc', 63 'shared_data.cc',
72 'shared_data.h', 64 'shared_data.h',
(...skipping 37 matching lines...) Expand 10 before | Expand all | Expand 10 after
110 }], 102 }],
111 ['include_tests==1', { 103 ['include_tests==1', {
112 'targets': [ 104 'targets': [
113 { 105 {
114 'target_name': 'voice_engine_unittests', 106 'target_name': 'voice_engine_unittests',
115 'type': '<(gtest_target_type)', 107 'type': '<(gtest_target_type)',
116 'dependencies': [ 108 'dependencies': [
117 'voice_engine', 109 'voice_engine',
118 '<(DEPTH)/testing/gmock.gyp:gmock', 110 '<(DEPTH)/testing/gmock.gyp:gmock',
119 '<(DEPTH)/testing/gtest.gyp:gtest', 111 '<(DEPTH)/testing/gtest.gyp:gtest',
120 '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
121 # The rest are to satisfy the unittests' include chain. 112 # The rest are to satisfy the unittests' include chain.
122 # This would be unnecessary if we used qualified includes. 113 # This would be unnecessary if we used qualified includes.
123 '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', 114 '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
124 '<(webrtc_root)/modules/modules.gyp:audio_device', 115 '<(webrtc_root)/modules/modules.gyp:audio_device',
125 '<(webrtc_root)/modules/modules.gyp:audio_processing', 116 '<(webrtc_root)/modules/modules.gyp:audio_processing',
126 '<(webrtc_root)/modules/modules.gyp:audio_coding_module', 117 '<(webrtc_root)/modules/modules.gyp:audio_coding_module',
127 '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer', 118 '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer',
128 '<(webrtc_root)/modules/modules.gyp:media_file', 119 '<(webrtc_root)/modules/modules.gyp:media_file',
129 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp', 120 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
130 '<(webrtc_root)/modules/modules.gyp:webrtc_utility', 121 '<(webrtc_root)/modules/modules.gyp:webrtc_utility',
131 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers' , 122 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers' ,
132 '<(webrtc_root)/test/test.gyp:test_support_main', 123 '<(webrtc_root)/test/test.gyp:test_support_main',
133 ], 124 ],
134 'sources': [ 125 'sources': [
135 'channel_unittest.cc', 126 'channel_unittest.cc',
136 'file_player_unittests.cc',
137 'network_predictor_unittest.cc', 127 'network_predictor_unittest.cc',
138 'transmit_mixer_unittest.cc', 128 'transmit_mixer_unittest.cc',
139 'utility_unittest.cc', 129 'utility_unittest.cc',
140 'voe_audio_processing_unittest.cc', 130 'voe_audio_processing_unittest.cc',
141 'voe_base_unittest.cc', 131 'voe_base_unittest.cc',
142 'voe_codec_unittest.cc', 132 'voe_codec_unittest.cc',
143 'voe_network_unittest.cc', 133 'voe_network_unittest.cc',
144 'voice_engine_fixture.cc', 134 'voice_engine_fixture.cc',
145 'voice_engine_fixture.h', 135 'voice_engine_fixture.h',
146 ], 136 ],
(...skipping 167 matching lines...) Expand 10 before | Expand all | Expand 10 after
314 'sources': [ 304 'sources': [
315 'voe_auto_test.isolate', 305 'voe_auto_test.isolate',
316 ], 306 ],
317 }, 307 },
318 ], 308 ],
319 }], 309 }],
320 ], # conditions 310 ], # conditions
321 }], # include_tests==1 311 }], # include_tests==1
322 ], # conditions 312 ], # conditions
323 } 313 }
OLDNEW
« no previous file with comments | « webrtc/voice_engine/transmit_mixer.h ('k') | no next file » | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698