webrtc_m130/webrtc/modules/audio_coding/main/audio_coding_module.gypi
henrik.lundin e9eca8f5ae Removing AudioCoding class, a.k.a the new ACM API
We have decided not to do a switch from old (AudioCodingModule) to new
(AudioCoding) API. Instead, we will gradually evolve the old API to
meet the new design goals.

As a consequence of this decision, the AudioCoding and AudioCodingImpl
classes are deleted. Also removing associated unit test sources. No
test coverage is lost with this operation, since the tests for the
"old" API are testing more than the deleted tests did.

BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1415163002

Cr-Commit-Position: refs/heads/master@{#10406}
2015-10-26 12:26:45 +00:00

173 lines
5.5 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.
{
'variables': {
'audio_coding_dependencies': [
'cng',
'g711',
'pcm16b',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
],
'audio_coding_defines': [],
'conditions': [
['include_opus==1', {
'audio_coding_dependencies': ['webrtc_opus',],
'audio_coding_defines': ['WEBRTC_CODEC_OPUS',],
}],
['build_with_mozilla==0', {
'conditions': [
['target_arch=="arm"', {
'audio_coding_dependencies': ['isac_fix',],
'audio_coding_defines': ['WEBRTC_CODEC_ISACFX',],
}, {
'audio_coding_dependencies': ['isac',],
'audio_coding_defines': ['WEBRTC_CODEC_ISAC',],
}],
],
'audio_coding_dependencies': ['g722',],
'audio_coding_defines': ['WEBRTC_CODEC_G722',],
}],
['build_with_mozilla==0 and build_with_chromium==0', {
'audio_coding_dependencies': ['ilbc', 'red',],
'audio_coding_defines': ['WEBRTC_CODEC_ILBC', 'WEBRTC_CODEC_RED',],
}],
],
},
'targets': [
{
'target_name': 'audio_coding_module',
'type': 'static_library',
'defines': [
'<@(audio_coding_defines)',
],
'dependencies': [
'<@(audio_coding_dependencies)',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/webrtc.gyp:rtc_event_log',
'neteq',
],
'include_dirs': [
'interface',
'../../interface',
'<(webrtc_root)',
],
'direct_dependent_settings': {
'include_dirs': [
'interface',
'../../interface',
'<(webrtc_root)',
],
},
'sources': [
'acm2/acm_codec_database.cc',
'acm2/acm_codec_database.h',
'acm2/acm_common_defs.h',
'acm2/acm_receiver.cc',
'acm2/acm_receiver.h',
'acm2/acm_resampler.cc',
'acm2/acm_resampler.h',
'acm2/audio_coding_module.cc',
'acm2/audio_coding_module_impl.cc',
'acm2/audio_coding_module_impl.h',
'acm2/call_statistics.cc',
'acm2/call_statistics.h',
'acm2/codec_manager.cc',
'acm2/codec_manager.h',
'acm2/codec_owner.cc',
'acm2/codec_owner.h',
'acm2/initial_delay_manager.cc',
'acm2/initial_delay_manager.h',
'acm2/nack.cc',
'acm2/nack.h',
'interface/audio_coding_module.h',
'interface/audio_coding_module_typedefs.h',
],
},
],
'conditions': [
['include_tests==1', {
'targets': [
{
'target_name': 'acm_receive_test',
'type': 'static_library',
'defines': [
'<@(audio_coding_defines)',
],
'dependencies': [
'<@(audio_coding_dependencies)',
'audio_coding_module',
'neteq_unittest_tools',
'<(DEPTH)/testing/gtest.gyp:gtest',
],
'sources': [
'acm2/acm_receive_test_oldapi.cc',
'acm2/acm_receive_test_oldapi.h',
],
}, # acm_receive_test
{
'target_name': 'acm_send_test',
'type': 'static_library',
'defines': [
'<@(audio_coding_defines)',
],
'dependencies': [
'<@(audio_coding_dependencies)',
'audio_coding_module',
'neteq_unittest_tools',
'<(DEPTH)/testing/gtest.gyp:gtest',
],
'sources': [
'acm2/acm_send_test_oldapi.cc',
'acm2/acm_send_test_oldapi.h',
],
}, # acm_send_test
{
'target_name': 'delay_test',
'type': 'executable',
'dependencies': [
'audio_coding_module',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/test/test.gyp:test_support',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
],
'sources': [
'test/delay_test.cc',
'test/Channel.cc',
'test/PCMFile.cc',
'test/utility.cc',
],
}, # delay_test
{
'target_name': 'insert_packet_with_timing',
'type': 'executable',
'dependencies': [
'audio_coding_module',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/test/test.gyp:test_support',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
],
'sources': [
'test/insert_packet_with_timing.cc',
'test/Channel.cc',
'test/PCMFile.cc',
],
}, # delay_test
],
}],
],
}