blob: 135d78b7510f8474b6806e139763361035072c3d [file] [log] [blame]
# Copyright (c) 2011 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': [
'../../..',
'../interface',
'<(webrtc_root)/modules/audio_device/main/source',
],
'direct_dependent_settings': {
'include_dirs': [
'../../..',
'../interface',
],
},
'sources': [
'../../../common_types.h',
'../../../engine_configurations.h',
'../../../typedefs.h',
'../interface/voe_audio_processing.h',
'../interface/voe_base.h',
'../interface/voe_call_report.h',
'../interface/voe_codec.h',
'../interface/voe_dtmf.h',
'../interface/voe_encryption.h',
'../interface/voe_errors.h',
'../interface/voe_external_media.h',
'../interface/voe_file.h',
'../interface/voe_hardware.h',
'../interface/voe_neteq_stats.h',
'../interface/voe_network.h',
'../interface/voe_rtp_rtcp.h',
'../interface/voe_video_sync.h',
'../interface/voe_volume_control.h',
'audio_frame_operations.cc',
'audio_frame_operations.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',
'ref_count.cc',
'ref_count.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': [
['build_with_chromium==0', {
'targets': [
{
'target_name': 'voice_engine_unittests',
'type': 'executable',
'dependencies': [
'voice_engine_core',
'<(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',
'<(webrtc_root)/../test/test.gyp:test_support_main',
'<(webrtc_root)/../testing/gtest.gyp:gtest',
],
'include_dirs': [
'../../..',
'../interface',
],
'sources': [
'channel_unittest.cc',
],
},
], # targets
}], # build_with_chromium
], # conditions
}
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