3fd7be4cb1
Reason for revert: Breaking Chromium FYI bots. Original issue's description: > Don't link with audio codecs that we don't use > > We used to link with all audio codecs unconditionally (except Opus); > this patch makes gyp and gn only link to the ones that are used. > > (This unfortunately fails to have a measurable impact on Chromium > binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC > fix were already being excluded from Chromium by some other means > (likely just the linker omitting compilation units with no incoming > references).) > > BUG=webrtc:4557 > > Committed: https://crrev.com/f66a9251424351ea6d631c54dd1feb64cc13d809 > Cr-Commit-Position: refs/heads/master@{#10046} TBR=henrik.lundin@webrtc.org,tina.legrand@webrtc.org,kjellander@webrtc.org,kwiberg@webrtc.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:4557 Review URL: https://codereview.webrtc.org/1368933002 Cr-Commit-Position: refs/heads/master@{#10069}
165 lines
5.1 KiB
Python
165 lines
5.1 KiB
Python
# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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{
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'variables': {
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'audio_coding_dependencies': [
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'cng',
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'g711',
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'g722',
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'ilbc',
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'isac',
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'isac_fix',
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'pcm16b',
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'red',
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'<(webrtc_root)/common.gyp:webrtc_common',
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'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
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'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
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],
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'audio_coding_defines': [],
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'conditions': [
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['include_opus==1', {
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'audio_coding_dependencies': ['webrtc_opus',],
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'audio_coding_defines': ['WEBRTC_CODEC_OPUS',],
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}],
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],
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},
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'targets': [
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{
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'target_name': 'audio_coding_module',
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'type': 'static_library',
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'defines': [
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'<@(audio_coding_defines)',
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],
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'dependencies': [
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'<@(audio_coding_dependencies)',
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'<(webrtc_root)/common.gyp:webrtc_common',
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'<(webrtc_root)/webrtc.gyp:rtc_event_log',
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'neteq',
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],
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'include_dirs': [
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'interface',
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'../../interface',
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'<(webrtc_root)',
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],
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'direct_dependent_settings': {
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'include_dirs': [
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'interface',
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'../../interface',
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'<(webrtc_root)',
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],
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},
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'sources': [
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'acm2/acm_codec_database.cc',
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'acm2/acm_codec_database.h',
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'acm2/acm_common_defs.h',
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'acm2/acm_receiver.cc',
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'acm2/acm_receiver.h',
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'acm2/acm_resampler.cc',
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'acm2/acm_resampler.h',
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'acm2/audio_coding_module.cc',
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'acm2/audio_coding_module_impl.cc',
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'acm2/audio_coding_module_impl.h',
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'acm2/call_statistics.cc',
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'acm2/call_statistics.h',
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'acm2/codec_manager.cc',
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'acm2/codec_manager.h',
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'acm2/codec_owner.cc',
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'acm2/codec_owner.h',
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'acm2/initial_delay_manager.cc',
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'acm2/initial_delay_manager.h',
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'acm2/nack.cc',
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'acm2/nack.h',
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'interface/audio_coding_module.h',
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'interface/audio_coding_module_typedefs.h',
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],
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},
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],
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'conditions': [
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['include_tests==1', {
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'targets': [
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{
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'target_name': 'acm_receive_test',
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'type': 'static_library',
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'defines': [
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'<@(audio_coding_defines)',
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],
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'dependencies': [
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'<@(audio_coding_dependencies)',
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'audio_coding_module',
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'neteq_unittest_tools',
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'<(DEPTH)/testing/gtest.gyp:gtest',
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],
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'sources': [
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'acm2/acm_receive_test.cc',
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'acm2/acm_receive_test.h',
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'acm2/acm_receive_test_oldapi.cc',
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'acm2/acm_receive_test_oldapi.h',
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],
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}, # acm_receive_test
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{
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'target_name': 'acm_send_test',
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'type': 'static_library',
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'defines': [
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'<@(audio_coding_defines)',
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],
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'dependencies': [
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'<@(audio_coding_dependencies)',
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'audio_coding_module',
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'neteq_unittest_tools',
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'<(DEPTH)/testing/gtest.gyp:gtest',
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],
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'sources': [
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'acm2/acm_send_test.cc',
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'acm2/acm_send_test.h',
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'acm2/acm_send_test_oldapi.cc',
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'acm2/acm_send_test_oldapi.h',
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],
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}, # acm_send_test
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{
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'target_name': 'delay_test',
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'type': 'executable',
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'dependencies': [
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'audio_coding_module',
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'<(DEPTH)/testing/gtest.gyp:gtest',
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'<(webrtc_root)/common.gyp:webrtc_common',
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'<(webrtc_root)/test/test.gyp:test_support',
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'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
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'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
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'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
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],
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'sources': [
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'test/delay_test.cc',
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'test/Channel.cc',
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'test/PCMFile.cc',
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'test/utility.cc',
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],
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}, # delay_test
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{
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'target_name': 'insert_packet_with_timing',
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'type': 'executable',
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'dependencies': [
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'audio_coding_module',
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'<(DEPTH)/testing/gtest.gyp:gtest',
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'<(webrtc_root)/common.gyp:webrtc_common',
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'<(webrtc_root)/test/test.gyp:test_support',
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'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
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'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
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'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
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],
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'sources': [
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'test/insert_packet_with_timing.cc',
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'test/Channel.cc',
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'test/PCMFile.cc',
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],
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}, # delay_test
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],
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}],
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],
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}
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