blob: 0478a710dbc58639fb85268ec85a0c230a684aa3 [file] [log] [blame]
andrew@webrtc.orga7b57da2012-10-22 18:19:23 +00001# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
2#
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
5# tree. An additional intellectual property rights grant can be found
6# in the file PATENTS. All contributing project authors may
7# be found in the AUTHORS file in the root of the source tree.
8
9{
10 'targets': [
11 {
12 'target_name': 'voice_engine_core',
13 'type': '<(library)',
14 'dependencies': [
15 '<(webrtc_root)/common_audio/common_audio.gyp:resampler',
16 '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
17 '<(webrtc_root)/modules/modules.gyp:audio_coding_module',
18 '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer',
19 '<(webrtc_root)/modules/modules.gyp:audio_device',
20 '<(webrtc_root)/modules/modules.gyp:audio_processing',
21 '<(webrtc_root)/modules/modules.gyp:media_file',
22 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
23 '<(webrtc_root)/modules/modules.gyp:udp_transport',
24 '<(webrtc_root)/modules/modules.gyp:webrtc_utility',
25 '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
26 ],
27 'include_dirs': [
28 'include',
29 '<(webrtc_root)/modules/audio_device',
30 ],
31 'direct_dependent_settings': {
32 'include_dirs': [
33 'include',
34 ],
35 },
36 'sources': [
37 '../common_types.h',
38 '../engine_configurations.h',
39 '../typedefs.h',
40 'include/voe_audio_processing.h',
41 'include/voe_base.h',
42 'include/voe_call_report.h',
43 'include/voe_codec.h',
44 'include/voe_dtmf.h',
45 'include/voe_encryption.h',
46 'include/voe_errors.h',
47 'include/voe_external_media.h',
48 'include/voe_file.h',
49 'include/voe_hardware.h',
50 'include/voe_neteq_stats.h',
51 'include/voe_network.h',
52 'include/voe_rtp_rtcp.h',
53 'include/voe_video_sync.h',
54 'include/voe_volume_control.h',
55 'channel.cc',
56 'channel.h',
57 'channel_manager.cc',
58 'channel_manager.h',
59 'channel_manager_base.cc',
60 'channel_manager_base.h',
61 'dtmf_inband.cc',
62 'dtmf_inband.h',
63 'dtmf_inband_queue.cc',
64 'dtmf_inband_queue.h',
65 'level_indicator.cc',
66 'level_indicator.h',
67 'monitor_module.cc',
68 'monitor_module.h',
69 'output_mixer.cc',
70 'output_mixer.h',
71 'output_mixer_internal.cc',
72 'output_mixer_internal.h',
73 'shared_data.cc',
74 'shared_data.h',
75 'statistics.cc',
76 'statistics.h',
77 'transmit_mixer.cc',
78 'transmit_mixer.h',
79 'utility.cc',
80 'utility.h',
81 'voe_audio_processing_impl.cc',
82 'voe_audio_processing_impl.h',
83 'voe_base_impl.cc',
84 'voe_base_impl.h',
85 'voe_call_report_impl.cc',
86 'voe_call_report_impl.h',
87 'voe_codec_impl.cc',
88 'voe_codec_impl.h',
89 'voe_dtmf_impl.cc',
90 'voe_dtmf_impl.h',
91 'voe_encryption_impl.cc',
92 'voe_encryption_impl.h',
93 'voe_external_media_impl.cc',
94 'voe_external_media_impl.h',
95 'voe_file_impl.cc',
96 'voe_file_impl.h',
97 'voe_hardware_impl.cc',
98 'voe_hardware_impl.h',
99 'voe_neteq_stats_impl.cc',
100 'voe_neteq_stats_impl.h',
101 'voe_network_impl.cc',
102 'voe_network_impl.h',
103 'voe_rtp_rtcp_impl.cc',
104 'voe_rtp_rtcp_impl.h',
105 'voe_video_sync_impl.cc',
106 'voe_video_sync_impl.h',
107 'voe_volume_control_impl.cc',
108 'voe_volume_control_impl.h',
109 'voice_engine_defines.h',
110 'voice_engine_impl.cc',
111 'voice_engine_impl.h',
112 ],
113 },
114 ],
115 'conditions': [
116 ['OS=="win"', {
117 'defines': ['WEBRTC_DRIFT_COMPENSATION_SUPPORTED',],
118 }],
119 ['include_tests==1', {
120 'targets': [
121 {
122 'target_name': 'voice_engine_unittests',
123 'type': 'executable',
124 'dependencies': [
125 'voice_engine_core',
126 '<(DEPTH)/testing/gtest.gyp:gtest',
127 '<(webrtc_root)/test/test.gyp:test_support_main',
128 # The rest are to satisfy the unittests' include chain.
129 # This would be unnecessary if we used qualified includes.
130 '<(webrtc_root)/common_audio/common_audio.gyp:resampler',
131 '<(webrtc_root)/modules/modules.gyp:audio_device',
132 '<(webrtc_root)/modules/modules.gyp:audio_processing',
133 '<(webrtc_root)/modules/modules.gyp:audio_coding_module',
134 '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer',
135 '<(webrtc_root)/modules/modules.gyp:media_file',
136 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
137 '<(webrtc_root)/modules/modules.gyp:udp_transport',
138 '<(webrtc_root)/modules/modules.gyp:webrtc_utility',
139 '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
140 ],
141 'include_dirs': [
142 'include',
143 ],
144 'sources': [
145 'channel_unittest.cc',
146 'output_mixer_unittest.cc',
147 'transmit_mixer_unittest.cc',
148 'voe_audio_processing_unittest.cc',
149 ],
150 },
151 ], # targets
152 }], # include_tests
153 ], # conditions
154}