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Rename voice_engine_core -> voice_engine and move targets to voice_engine.gyp.

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R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1574004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4126 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org 2013-05-29 00:38:02 +00:00
parent a6ae644e52
commit f791b1cebf
10 changed files with 300 additions and 333 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
tools/e2e_quality/e2e_quality.gyp
View File

@ -14,7 +14,7 @@
'type': 'executable',
'dependencies': [
'<(webrtc_root)/test/channel_transport.gyp:channel_transport',
'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine_core',
'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/third_party/google-gflags/google-gflags.gyp:google-gflags',
],

2
webrtc/build/merge_voice_libs.gyp
View File

@ -13,7 +13,7 @@
'target_name': 'no_op',
'type': 'executable',
'dependencies': [
'../voice_engine/voice_engine.gyp:voice_engine_core',
'../voice_engine/voice_engine.gyp:voice_engine',
],
'sources': [ 'no_op.cc', ],
},

2
webrtc/tools/tools.gyp
View File

@ -98,7 +98,7 @@
'target_name': 'force_mic_volume_max',
'type': 'executable',
'dependencies': [
'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine_core',
'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
],
'sources': [
'force_mic_volume_max/force_mic_volume_max.cc',

2
webrtc/video_engine/test/android/android_video_demo.gypi
View File

@ -14,7 +14,7 @@
'<(webrtc_root)/modules/modules.gyp:*',
'<(webrtc_root)/test/channel_transport.gyp:channel_transport',
'<(webrtc_root)/video_engine/video_engine.gyp:video_engine_core',
'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine_core',
'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
],
'actions': [
{

6
webrtc/video_engine/test/android/jni/Android.mk
View File

@ -16,9 +16,9 @@ BUILDTYPE ?= Debug
MY_LIBS_PATH := ../../../../../out/$(BUILDTYPE)
include $(CLEAR_VARS)
LOCAL_MODULE := libvoice_engine_core
LOCAL_MODULE := libvoice_engine
LOCAL_SRC_FILES := \
$(MY_LIBS_PATH)/libvoice_engine_core.a
$(MY_LIBS_PATH)/libvoice_engine.a
include $(PREBUILT_STATIC_LIBRARY)
include $(CLEAR_VARS)
@ -360,7 +360,7 @@ LOCAL_LDLIBS := \
# LOCAL_STATIC_LIBRARIES after removing the circular dependencies between
# libvpx.a and libvpx_intrinsics_[mmx,sse2,ssse3].a
LOCAL_WHOLE_STATIC_LIBRARIES := \
libvoice_engine_core \
libvoice_engine \
libvideo_engine_core \
libvideo_processing \
libwebrtc_video_coding \

2
webrtc/video_engine/test/auto_test/vie_auto_test.gypi
View File

@ -15,7 +15,7 @@
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/modules/modules.gyp:video_render_module',
'<(webrtc_root)/modules/modules.gyp:video_capture_module',
'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine_core',
'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/third_party/google-gflags/google-gflags.gyp:google-gflags',
'<(webrtc_root)/test/metrics.gyp:metrics',

2
webrtc/video_engine/video_engine_core.gypi
View File

@ -29,7 +29,7 @@
'<(webrtc_root)/modules/modules.gyp:video_render_module',
# VoiceEngine
'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine_core',
'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
# system_wrappers
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',

163
webrtc/voice_engine/test/voice_engine_tests.gypi
View File

@ -1,163 +0,0 @@
# 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': [
# Auto test - command line test for all platforms
{
'target_name': 'voe_auto_test',
'type': 'executable',
'dependencies': [
'voice_engine_core',
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/test/test.gyp:test_support',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/testing/gmock.gyp:gmock',
'<(DEPTH)/third_party/google-gflags/google-gflags.gyp:google-gflags',
'<(webrtc_root)/test/libtest/libtest.gyp:libtest',
'<(webrtc_root)/test/channel_transport.gyp:channel_transport',
],
'include_dirs': [
'auto_test',
'auto_test/fixtures',
'<(webrtc_root)/modules/interface',
# TODO(phoglund): We only depend on voice_engine_defines.h here -
# move that file to interface and then remove this dependency.
'<(webrtc_root)/voice_engine',
'<(webrtc_root)/modules/audio_device/main/interface',
],
'sources': [
'auto_test/automated_mode.cc',
'auto_test/extended/agc_config_test.cc',
'auto_test/extended/ec_metrics_test.cc',
'auto_test/fakes/fake_external_transport.cc',
'auto_test/fakes/fake_external_transport.h',
'auto_test/fixtures/after_initialization_fixture.cc',
'auto_test/fixtures/after_initialization_fixture.h',
'auto_test/fixtures/after_streaming_fixture.cc',
'auto_test/fixtures/after_streaming_fixture.h',
'auto_test/fixtures/before_initialization_fixture.cc',
'auto_test/fixtures/before_initialization_fixture.h',
'auto_test/fuzz/rtp_fuzz_test.cc',
'auto_test/standard/audio_processing_test.cc',
'auto_test/standard/call_report_test.cc',
'auto_test/standard/codec_before_streaming_test.cc',
'auto_test/standard/codec_test.cc',
'auto_test/standard/dtmf_test.cc',
'auto_test/standard/encryption_test.cc',
'auto_test/standard/external_media_test.cc',
'auto_test/standard/file_before_streaming_test.cc',
'auto_test/standard/file_test.cc',
'auto_test/standard/hardware_before_initializing_test.cc',
'auto_test/standard/hardware_before_streaming_test.cc',
'auto_test/standard/hardware_test.cc',
'auto_test/standard/manual_hold_test.cc',
'auto_test/standard/mixing_test.cc',
'auto_test/standard/neteq_stats_test.cc',
'auto_test/standard/neteq_test.cc',
'auto_test/standard/network_test.cc',
'auto_test/standard/rtp_rtcp_before_streaming_test.cc',
'auto_test/standard/rtp_rtcp_test.cc',
'auto_test/standard/voe_base_misc_test.cc',
'auto_test/standard/video_sync_test.cc',
'auto_test/standard/volume_test.cc',
'auto_test/resource_manager.cc',
'auto_test/voe_cpu_test.cc',
'auto_test/voe_cpu_test.h',
'auto_test/voe_extended_test.cc',
'auto_test/voe_extended_test.h',
'auto_test/voe_standard_test.cc',
'auto_test/voe_standard_test.h',
'auto_test/voe_stress_test.cc',
'auto_test/voe_stress_test.h',
'auto_test/voe_test_defines.h',
'auto_test/voe_test_interface.h',
'auto_test/voe_unit_test.cc',
'auto_test/voe_unit_test.h',
],
'conditions': [
['OS=="android"', {
# some tests are not supported on android yet, exclude these tests.
'sources!': [
'auto_test/standard/hardware_before_streaming_test.cc',
],
}],
],
# Disable warnings to enable Win64 build, issue 1323.
'msvs_disabled_warnings': [
4267, # size_t to int truncation.
],
},
{
# command line test that should work on linux/mac/win
'target_name': 'voe_cmd_test',
'type': 'executable',
'dependencies': [
'<(webrtc_root)/test/test.gyp:test_support',
'<(DEPTH)/testing/gtest.gyp:gtest',
'voice_engine_core',
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/test/channel_transport.gyp:channel_transport',
],
'sources': [
'cmd_test/voe_cmd_test.cc',
],
},
],
'conditions': [
# TODO(kjellander): Support UseoFMFC on VS2010.
# http://code.google.com/p/webrtc/issues/detail?id=709
['OS=="win" and MSVS_VERSION < "2010"', {
'targets': [
# WinTest - GUI test for Windows
{
'target_name': 'voe_ui_win_test',
'type': 'executable',
'dependencies': [
'voice_engine_core',
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/test/test.gyp:test_support',
],
'include_dirs': [
'win_test',
],
'sources': [
'win_test/Resource.h',
'win_test/WinTest.cc',
'win_test/WinTest.h',
'win_test/WinTest.rc',
'win_test/WinTestDlg.cc',
'win_test/WinTestDlg.h',
'win_test/res/WinTest.ico',
'win_test/res/WinTest.rc2',
'win_test/stdafx.cc',
'win_test/stdafx.h',
],
'configurations': {
'Common_Base': {
'msvs_configuration_attributes': {
'conditions': [
['component=="shared_library"', {
'UseOfMFC': '2', # Shared DLL
},{
'UseOfMFC': '1', # Static
}],
],
},
},
},
'msvs_settings': {
'VCLinkerTool': {
'SubSystem': '2', # Windows
},
},
},
],
}],
],
}

299
webrtc/voice_engine/voice_engine.gyp
View File

@ -9,15 +9,298 @@
{
'includes': [
'../build/common.gypi',
'voice_engine_core.gypi',
],
# Test targets, excluded when building with Chromium.
'conditions': [
['include_tests==1', {
'includes': [
'test/voice_engine_tests.gypi',
'targets': [
{
'target_name': 'voice_engine',
'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',
'<(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',
],
},
{
'target_name': 'voe_auto_test',
'type': 'executable',
'dependencies': [
'voice_engine',
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/test/test.gyp:test_support',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/testing/gmock.gyp:gmock',
'<(DEPTH)/third_party/google-gflags/google-gflags.gyp:google-gflags',
'<(webrtc_root)/test/libtest/libtest.gyp:libtest',
'<(webrtc_root)/test/channel_transport.gyp:channel_transport',
],
'include_dirs': [
'auto_test',
'auto_test/fixtures',
'<(webrtc_root)/modules/interface',
# TODO(phoglund): We only depend on voice_engine_defines.h here -
# move that file to interface and then remove this dependency.
'<(webrtc_root)/voice_engine',
'<(webrtc_root)/modules/audio_device/main/interface',
],
'sources': [
'test/auto_test/automated_mode.cc',
'test/auto_test/extended/agc_config_test.cc',
'test/auto_test/extended/ec_metrics_test.cc',
'test/auto_test/fakes/fake_external_transport.cc',
'test/auto_test/fakes/fake_external_transport.h',
'test/auto_test/fixtures/after_initialization_fixture.cc',
'test/auto_test/fixtures/after_initialization_fixture.h',
'test/auto_test/fixtures/after_streaming_fixture.cc',
'test/auto_test/fixtures/after_streaming_fixture.h',
'test/auto_test/fixtures/before_initialization_fixture.cc',
'test/auto_test/fixtures/before_initialization_fixture.h',
'test/auto_test/fuzz/rtp_fuzz_test.cc',
'test/auto_test/standard/audio_processing_test.cc',
'test/auto_test/standard/call_report_test.cc',
'test/auto_test/standard/codec_before_streaming_test.cc',
'test/auto_test/standard/codec_test.cc',
'test/auto_test/standard/dtmf_test.cc',
'test/auto_test/standard/encryption_test.cc',
'test/auto_test/standard/external_media_test.cc',
'test/auto_test/standard/file_before_streaming_test.cc',
'test/auto_test/standard/file_test.cc',
'test/auto_test/standard/hardware_before_initializing_test.cc',
'test/auto_test/standard/hardware_before_streaming_test.cc',
'test/auto_test/standard/hardware_test.cc',
'test/auto_test/standard/manual_hold_test.cc',
'test/auto_test/standard/mixing_test.cc',
'test/auto_test/standard/neteq_stats_test.cc',
'test/auto_test/standard/neteq_test.cc',
'test/auto_test/standard/network_test.cc',
'test/auto_test/standard/rtp_rtcp_before_streaming_test.cc',
'test/auto_test/standard/rtp_rtcp_test.cc',
'test/auto_test/standard/voe_base_misc_test.cc',
'test/auto_test/standard/video_sync_test.cc',
'test/auto_test/standard/volume_test.cc',
'test/auto_test/resource_manager.cc',
'test/auto_test/voe_cpu_test.cc',
'test/auto_test/voe_cpu_test.h',
'test/auto_test/voe_extended_test.cc',
'test/auto_test/voe_extended_test.h',
'test/auto_test/voe_standard_test.cc',
'test/auto_test/voe_standard_test.h',
'test/auto_test/voe_stress_test.cc',
'test/auto_test/voe_stress_test.h',
'test/auto_test/voe_test_defines.h',
'test/auto_test/voe_test_interface.h',
'test/auto_test/voe_unit_test.cc',
'test/auto_test/voe_unit_test.h',
],
'conditions': [
['OS=="android"', {
# some tests are not supported on android yet, exclude these tests.
'sources!': [
'test/auto_test/standard/hardware_before_streaming_test.cc',
],
}],
],
# Disable warnings to enable Win64 build, issue 1323.
'msvs_disabled_warnings': [
4267, # size_t to int truncation.
],
},
{
# command line test that should work on linux/mac/win
'target_name': 'voe_cmd_test',
'type': 'executable',
'dependencies': [
'<(webrtc_root)/test/test.gyp:test_support',
'<(DEPTH)/testing/gtest.gyp:gtest',
'voice_engine',
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/test/channel_transport.gyp:channel_transport',
],
'sources': [
'test/cmd_test/voe_cmd_test.cc',
],
},
], # targets
'conditions': [
# TODO(kjellander): Support UseoFMFC on VS2010.
# http://code.google.com/p/webrtc/issues/detail?id=709
['OS=="win" and MSVS_VERSION < "2010"', {
'targets': [
# WinTest - GUI test for Windows
{
'target_name': 'voe_ui_win_test',
'type': 'executable',
'dependencies': [
'voice_engine',
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/test/test.gyp:test_support',
],
'include_dirs': [
'win_test',
],
'sources': [
'test/win_test/Resource.h',
'test/win_test/WinTest.cc',
'test/win_test/WinTest.h',
'test/win_test/WinTest.rc',
'test/win_test/WinTestDlg.cc',
'test/win_test/WinTestDlg.h',
'test/win_test/res/WinTest.ico',
'test/win_test/res/WinTest.rc2',
'test/win_test/stdafx.cc',
'test/win_test/stdafx.h',
],
'configurations': {
'Common_Base': {
'msvs_configuration_attributes': {
'conditions': [
['component=="shared_library"', {
'UseOfMFC': '2', # Shared DLL
},{
'UseOfMFC': '1', # Static
}],
],
},
},
},
'msvs_settings': {
'VCLinkerTool': {
'SubSystem': '2', # Windows
},
},
},
], # targets
}],
], # conditions
}], # include_tests
], # conditions
}

153
webrtc/voice_engine/voice_engine_core.gypi
View File

@ -1,153 +0,0 @@
# 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
}