| # 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': '<(library)', |
| 'dependencies': [ |
| '<(webrtc_root)/common_audio/common_audio.gyp:resampler', |
| '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing', |
| '<(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:udp_transport', |
| '<(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:resampler', |
| '<(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:udp_transport', |
| '<(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', |
| ], |
| }, |
| ], # targets |
| }], # include_tests |
| ], # conditions |
| } |