ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
194 lines
6.0 KiB
Python
194 lines
6.0 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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'pcm16b',
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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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['build_with_mozilla==0', {
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'conditions': [
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['target_arch=="arm"', {
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'audio_coding_dependencies': ['isac_fix',],
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'audio_coding_defines': ['WEBRTC_CODEC_ISACFX',],
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}, {
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'audio_coding_dependencies': ['isac',],
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'audio_coding_defines': ['WEBRTC_CODEC_ISAC',],
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}],
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],
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'audio_coding_dependencies': ['g722',],
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'audio_coding_defines': ['WEBRTC_CODEC_G722',],
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}],
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['build_with_mozilla==0 and build_with_chromium==0', {
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'audio_coding_dependencies': ['ilbc', 'red',],
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'audio_coding_defines': ['WEBRTC_CODEC_ILBC', 'WEBRTC_CODEC_RED',],
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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': 'rent_a_codec',
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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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'<(webrtc_root)/common.gyp:webrtc_common',
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],
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'include_dirs': [
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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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'<(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/rent_a_codec.cc',
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'acm2/rent_a_codec.h',
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],
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},
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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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'rent_a_codec',
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],
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'include_dirs': [
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'include',
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'../../include',
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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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'include',
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'../../include',
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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_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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'include/audio_coding_module.h',
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'include/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_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_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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