342353780d
In principle should reduce gyp processing time, but the difference was not measurable. In any case, it's a good simplification that aligns with having a single common_video target. R=bjornv@webrtc.org, kma@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1375004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3928 4adac7df-926f-26a2-2b94-8c16560cd09d
154 lines
5.2 KiB
Python
154 lines
5.2 KiB
Python
# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
|
|
#
|
|
# Use of this source code is governed by a BSD-style license
|
|
# that can be found in the LICENSE file in the root of the source
|
|
# tree. An additional intellectual property rights grant can be found
|
|
# in the file PATENTS. All contributing project authors may
|
|
# be found in the AUTHORS file in the root of the source tree.
|
|
|
|
{
|
|
'targets': [
|
|
{
|
|
'target_name': 'voice_engine_core',
|
|
'type': 'static_library',
|
|
'dependencies': [
|
|
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
|
|
'<(webrtc_root)/modules/modules.gyp:audio_coding_module',
|
|
'<(webrtc_root)/modules/modules.gyp:audio_conference_mixer',
|
|
'<(webrtc_root)/modules/modules.gyp:audio_device',
|
|
'<(webrtc_root)/modules/modules.gyp:audio_processing',
|
|
'<(webrtc_root)/modules/modules.gyp:media_file',
|
|
'<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
|
|
'<(webrtc_root)/modules/modules.gyp:webrtc_utility',
|
|
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
|
|
],
|
|
'include_dirs': [
|
|
'include',
|
|
'<(webrtc_root)/modules/audio_device',
|
|
],
|
|
'direct_dependent_settings': {
|
|
'include_dirs': [
|
|
'include',
|
|
],
|
|
},
|
|
'sources': [
|
|
'../common_types.h',
|
|
'../engine_configurations.h',
|
|
'../typedefs.h',
|
|
'include/voe_audio_processing.h',
|
|
'include/voe_base.h',
|
|
'include/voe_call_report.h',
|
|
'include/voe_codec.h',
|
|
'include/voe_dtmf.h',
|
|
'include/voe_encryption.h',
|
|
'include/voe_errors.h',
|
|
'include/voe_external_media.h',
|
|
'include/voe_file.h',
|
|
'include/voe_hardware.h',
|
|
'include/voe_neteq_stats.h',
|
|
'include/voe_network.h',
|
|
'include/voe_rtp_rtcp.h',
|
|
'include/voe_video_sync.h',
|
|
'include/voe_volume_control.h',
|
|
'channel.cc',
|
|
'channel.h',
|
|
'channel_manager.cc',
|
|
'channel_manager.h',
|
|
'channel_manager_base.cc',
|
|
'channel_manager_base.h',
|
|
'dtmf_inband.cc',
|
|
'dtmf_inband.h',
|
|
'dtmf_inband_queue.cc',
|
|
'dtmf_inband_queue.h',
|
|
'level_indicator.cc',
|
|
'level_indicator.h',
|
|
'monitor_module.cc',
|
|
'monitor_module.h',
|
|
'output_mixer.cc',
|
|
'output_mixer.h',
|
|
'output_mixer_internal.cc',
|
|
'output_mixer_internal.h',
|
|
'shared_data.cc',
|
|
'shared_data.h',
|
|
'statistics.cc',
|
|
'statistics.h',
|
|
'transmit_mixer.cc',
|
|
'transmit_mixer.h',
|
|
'utility.cc',
|
|
'utility.h',
|
|
'voe_audio_processing_impl.cc',
|
|
'voe_audio_processing_impl.h',
|
|
'voe_base_impl.cc',
|
|
'voe_base_impl.h',
|
|
'voe_call_report_impl.cc',
|
|
'voe_call_report_impl.h',
|
|
'voe_codec_impl.cc',
|
|
'voe_codec_impl.h',
|
|
'voe_dtmf_impl.cc',
|
|
'voe_dtmf_impl.h',
|
|
'voe_encryption_impl.cc',
|
|
'voe_encryption_impl.h',
|
|
'voe_external_media_impl.cc',
|
|
'voe_external_media_impl.h',
|
|
'voe_file_impl.cc',
|
|
'voe_file_impl.h',
|
|
'voe_hardware_impl.cc',
|
|
'voe_hardware_impl.h',
|
|
'voe_neteq_stats_impl.cc',
|
|
'voe_neteq_stats_impl.h',
|
|
'voe_network_impl.cc',
|
|
'voe_network_impl.h',
|
|
'voe_rtp_rtcp_impl.cc',
|
|
'voe_rtp_rtcp_impl.h',
|
|
'voe_video_sync_impl.cc',
|
|
'voe_video_sync_impl.h',
|
|
'voe_volume_control_impl.cc',
|
|
'voe_volume_control_impl.h',
|
|
'voice_engine_defines.h',
|
|
'voice_engine_impl.cc',
|
|
'voice_engine_impl.h',
|
|
],
|
|
},
|
|
],
|
|
'conditions': [
|
|
['OS=="win"', {
|
|
'defines': ['WEBRTC_DRIFT_COMPENSATION_SUPPORTED',],
|
|
}],
|
|
['include_tests==1', {
|
|
'targets': [
|
|
{
|
|
'target_name': 'voice_engine_unittests',
|
|
'type': 'executable',
|
|
'dependencies': [
|
|
'voice_engine_core',
|
|
'<(DEPTH)/testing/gtest.gyp:gtest',
|
|
'<(webrtc_root)/test/test.gyp:test_support_main',
|
|
# The rest are to satisfy the unittests' include chain.
|
|
# This would be unnecessary if we used qualified includes.
|
|
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
|
|
'<(webrtc_root)/modules/modules.gyp:audio_device',
|
|
'<(webrtc_root)/modules/modules.gyp:audio_processing',
|
|
'<(webrtc_root)/modules/modules.gyp:audio_coding_module',
|
|
'<(webrtc_root)/modules/modules.gyp:audio_conference_mixer',
|
|
'<(webrtc_root)/modules/modules.gyp:media_file',
|
|
'<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
|
|
'<(webrtc_root)/modules/modules.gyp:webrtc_utility',
|
|
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
|
|
],
|
|
'include_dirs': [
|
|
'include',
|
|
],
|
|
'sources': [
|
|
'channel_unittest.cc',
|
|
'output_mixer_unittest.cc',
|
|
'transmit_mixer_unittest.cc',
|
|
'voe_audio_processing_unittest.cc',
|
|
'voe_base_unittest.cc',
|
|
'voe_codec_unittest.cc',
|
|
],
|
|
},
|
|
], # targets
|
|
}], # include_tests
|
|
], # conditions
|
|
}
|