From 7056be937f040e8d794b6f733e1eb409a62dceef Mon Sep 17 00:00:00 2001 From: mflodman Date: Fri, 7 Oct 2016 07:07:28 +0200 Subject: [PATCH] Delete old video defines in engine config. This CL deletes the old and not used video defines in engine_configurations.h and pre-pends voice_ to indicate there are only voice/audio defines left in the file. BUG=none R=solenberg@webrtc.org Review URL: https://codereview.webrtc.org/2401673002 . Cr-Commit-Position: refs/heads/master@{#14558} --- webrtc/BUILD.gn | 2 +- webrtc/DEPS | 2 +- webrtc/common.gyp | 2 +- .../audio_coding/acm2/acm_codec_database.h | 2 +- .../audio_coding/acm2/acm_common_defs.h | 2 +- .../modules/audio_coding/acm2/acm_receiver.h | 2 +- .../audio_coding/acm2/codec_manager.cc | 2 +- .../audio_coding/neteq/audio_decoder_impl.h | 2 +- webrtc/modules/audio_coding/test/APITest.cc | 2 +- webrtc/modules/audio_coding/test/RTPFile.cc | 2 +- .../audio_coding/test/TestAllCodecs.cc | 2 +- .../modules/audio_coding/test/TestRedFec.cc | 2 +- .../modules/audio_coding/test/TestStereo.cc | 2 +- .../modules/audio_coding/test/TestVADDTX.cc | 2 +- .../audio_coding/test/TwoWayCommunication.cc | 2 +- .../modules/audio_coding/test/delay_test.cc | 2 +- webrtc/modules/audio_coding/test/opus_test.cc | 2 +- .../source/audio_conference_mixer_impl.h | 2 +- webrtc/modules/audio_mixer/audio_mixer_impl.h | 2 +- .../modules/media_file/media_file_defines.h | 2 +- .../modules/media_file/media_file_utility.cc | 2 +- webrtc/modules/utility/include/file_player.h | 2 +- .../modules/utility/include/file_recorder.h | 2 +- webrtc/modules/utility/source/file_player.cc | 2 +- .../modules/utility/source/file_recorder.cc | 2 +- webrtc/modules/video_coding/codec_database.cc | 1 - .../modules/video_coding/generic_encoder.cc | 1 - webrtc/video/rtp_stream_receiver.h | 1 - .../fixtures/before_initialization_fixture.h | 2 +- .../test/auto_test/voe_standard_test.cc | 2 +- .../test/auto_test/voe_test_common.h | 2 +- .../test/auto_test/voe_test_defines.h | 4 +-- .../test/cmd_test/voe_cmd_test.cc | 2 +- webrtc/voice_engine/voice_engine_defines.h | 2 +- webrtc/voice_engine/voice_engine_impl.h | 2 +- ...ations.h => voice_engine_configurations.h} | 34 ++----------------- 36 files changed, 36 insertions(+), 67 deletions(-) rename webrtc/{engine_configurations.h => voice_engine_configurations.h} (60%) diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn index d47cdf9271..619f1d3d00 100644 --- a/webrtc/BUILD.gn +++ b/webrtc/BUILD.gn @@ -298,8 +298,8 @@ rtc_static_library("webrtc_common") { "common_types.h", "config.cc", "config.h", - "engine_configurations.h", "typedefs.h", + "voice_engine_configurations.h", ] if (is_clang && !is_nacl) { diff --git a/webrtc/DEPS b/webrtc/DEPS index 27b0eecea4..cee05ef106 100644 --- a/webrtc/DEPS +++ b/webrtc/DEPS @@ -14,7 +14,6 @@ include_rules = [ "+webrtc/common_types.h", "+webrtc/common_video/rotation.h", "+webrtc/config.h", - "+webrtc/engine_configurations.h", "+webrtc/transport.h", "+webrtc/typedefs.h", "+webrtc/video_decoder.h", @@ -23,6 +22,7 @@ include_rules = [ "+webrtc/video_receive_stream.h", "+webrtc/video_renderer.h", "+webrtc/video_send_stream.h", + "+webrtc/voice_engine_configurations.h", "+WebRTC", "+webrtc/api", diff --git a/webrtc/common.gyp b/webrtc/common.gyp index 3b5fe902dd..ad3f9f2285 100644 --- a/webrtc/common.gyp +++ b/webrtc/common.gyp @@ -16,8 +16,8 @@ 'common_types.h', 'config.h', 'config.cc', - 'engine_configurations.h', 'typedefs.h', + 'voice_engine_configurations.h', ], }, ], diff --git a/webrtc/modules/audio_coding/acm2/acm_codec_database.h b/webrtc/modules/audio_coding/acm2/acm_codec_database.h index 6c2db9cfc8..98569dd7bc 100644 --- a/webrtc/modules/audio_coding/acm2/acm_codec_database.h +++ b/webrtc/modules/audio_coding/acm2/acm_codec_database.h @@ -17,9 +17,9 @@ #define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_CODEC_DATABASE_H_ #include "webrtc/common_types.h" -#include "webrtc/engine_configurations.h" #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" #include "webrtc/modules/audio_coding/neteq/include/neteq.h" +#include "webrtc/voice_engine_configurations.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/acm2/acm_common_defs.h b/webrtc/modules/audio_coding/acm2/acm_common_defs.h index 483bdd93f1..e708978632 100644 --- a/webrtc/modules/audio_coding/acm2/acm_common_defs.h +++ b/webrtc/modules/audio_coding/acm2/acm_common_defs.h @@ -11,7 +11,7 @@ #ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_COMMON_DEFS_H_ #define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_COMMON_DEFS_H_ -#include "webrtc/engine_configurations.h" +#include "webrtc/voice_engine_configurations.h" // Checks for enabled codecs, we prevent enabling codecs which are not // compatible. diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.h b/webrtc/modules/audio_coding/acm2/acm_receiver.h index ea85456458..a9550fbf41 100644 --- a/webrtc/modules/audio_coding/acm2/acm_receiver.h +++ b/webrtc/modules/audio_coding/acm2/acm_receiver.h @@ -21,13 +21,13 @@ #include "webrtc/base/optional.h" #include "webrtc/base/thread_annotations.h" #include "webrtc/common_audio/vad/include/webrtc_vad.h" -#include "webrtc/engine_configurations.h" #include "webrtc/modules/audio_coding/acm2/acm_resampler.h" #include "webrtc/modules/audio_coding/acm2/call_statistics.h" #include "webrtc/modules/audio_coding/include/audio_coding_module.h" #include "webrtc/modules/audio_coding/neteq/include/neteq.h" #include "webrtc/modules/include/module_common_types.h" #include "webrtc/typedefs.h" +#include "webrtc/voice_engine_configurations.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/acm2/codec_manager.cc b/webrtc/modules/audio_coding/acm2/codec_manager.cc index f028c45f99..d8dcd792c4 100644 --- a/webrtc/modules/audio_coding/acm2/codec_manager.cc +++ b/webrtc/modules/audio_coding/acm2/codec_manager.cc @@ -12,9 +12,9 @@ #include "webrtc/base/checks.h" #include "webrtc/base/format_macros.h" -#include "webrtc/engine_configurations.h" #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" #include "webrtc/system_wrappers/include/trace.h" +#include "webrtc/voice_engine_configurations.h" namespace webrtc { namespace acm2 { diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h index 8636922509..559664433a 100644 --- a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h +++ b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h @@ -13,7 +13,6 @@ #include -#include "webrtc/engine_configurations.h" #include "webrtc/base/constructormagic.h" #include "webrtc/modules/audio_coding/codecs/audio_decoder.h" #ifdef WEBRTC_CODEC_G722 @@ -21,6 +20,7 @@ #endif #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" #include "webrtc/typedefs.h" +#include "webrtc/voice_engine_configurations.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/test/APITest.cc b/webrtc/modules/audio_coding/test/APITest.cc index 717de82ff6..ffa675c0fa 100644 --- a/webrtc/modules/audio_coding/test/APITest.cc +++ b/webrtc/modules/audio_coding/test/APITest.cc @@ -22,13 +22,13 @@ #include "webrtc/base/platform_thread.h" #include "webrtc/base/timeutils.h" #include "webrtc/common_types.h" -#include "webrtc/engine_configurations.h" #include "webrtc/modules/audio_coding/acm2/acm_common_defs.h" #include "webrtc/modules/audio_coding/test/utility.h" #include "webrtc/system_wrappers/include/event_wrapper.h" #include "webrtc/system_wrappers/include/trace.h" #include "webrtc/test/gtest.h" #include "webrtc/test/testsupport/fileutils.h" +#include "webrtc/voice_engine_configurations.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/test/RTPFile.cc b/webrtc/modules/audio_coding/test/RTPFile.cc index d59d38cef4..a61d0b0392 100644 --- a/webrtc/modules/audio_coding/test/RTPFile.cc +++ b/webrtc/modules/audio_coding/test/RTPFile.cc @@ -20,10 +20,10 @@ #endif #include "audio_coding_module.h" -#include "webrtc/engine_configurations.h" #include "webrtc/system_wrappers/include/rw_lock_wrapper.h" // TODO(tlegrand): Consider removing usage of gtest. #include "webrtc/test/gtest.h" +#include "webrtc/voice_engine_configurations.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/test/TestAllCodecs.cc b/webrtc/modules/audio_coding/test/TestAllCodecs.cc index 96c359c6d5..e5d9784e9c 100644 --- a/webrtc/modules/audio_coding/test/TestAllCodecs.cc +++ b/webrtc/modules/audio_coding/test/TestAllCodecs.cc @@ -15,7 +15,6 @@ #include #include "webrtc/common_types.h" -#include "webrtc/engine_configurations.h" #include "webrtc/modules/audio_coding/include/audio_coding_module.h" #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h" #include "webrtc/modules/audio_coding/test/utility.h" @@ -23,6 +22,7 @@ #include "webrtc/test/gtest.h" #include "webrtc/test/testsupport/fileutils.h" #include "webrtc/typedefs.h" +#include "webrtc/voice_engine_configurations.h" // Description of the test: // In this test we set up a one-way communication channel from a participant diff --git a/webrtc/modules/audio_coding/test/TestRedFec.cc b/webrtc/modules/audio_coding/test/TestRedFec.cc index dbb238f499..4e8b123b78 100644 --- a/webrtc/modules/audio_coding/test/TestRedFec.cc +++ b/webrtc/modules/audio_coding/test/TestRedFec.cc @@ -13,11 +13,11 @@ #include #include "webrtc/common_types.h" -#include "webrtc/engine_configurations.h" #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h" #include "webrtc/modules/audio_coding/test/utility.h" #include "webrtc/system_wrappers/include/trace.h" #include "webrtc/test/testsupport/fileutils.h" +#include "webrtc/voice_engine_configurations.h" #ifdef SUPPORT_RED_WB #undef SUPPORT_RED_WB diff --git a/webrtc/modules/audio_coding/test/TestStereo.cc b/webrtc/modules/audio_coding/test/TestStereo.cc index abc61796ee..572a08c21e 100644 --- a/webrtc/modules/audio_coding/test/TestStereo.cc +++ b/webrtc/modules/audio_coding/test/TestStereo.cc @@ -15,12 +15,12 @@ #include #include "webrtc/common_types.h" -#include "webrtc/engine_configurations.h" #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h" #include "webrtc/modules/audio_coding/test/utility.h" #include "webrtc/system_wrappers/include/trace.h" #include "webrtc/test/gtest.h" #include "webrtc/test/testsupport/fileutils.h" +#include "webrtc/voice_engine_configurations.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/test/TestVADDTX.cc b/webrtc/modules/audio_coding/test/TestVADDTX.cc index 541dfc3c45..505eb13285 100644 --- a/webrtc/modules/audio_coding/test/TestVADDTX.cc +++ b/webrtc/modules/audio_coding/test/TestVADDTX.cc @@ -12,10 +12,10 @@ #include -#include "webrtc/engine_configurations.h" #include "webrtc/modules/audio_coding/test/PCMFile.h" #include "webrtc/modules/audio_coding/test/utility.h" #include "webrtc/test/testsupport/fileutils.h" +#include "webrtc/voice_engine_configurations.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/test/TwoWayCommunication.cc b/webrtc/modules/audio_coding/test/TwoWayCommunication.cc index 9c26a32c91..ca2a2124ae 100644 --- a/webrtc/modules/audio_coding/test/TwoWayCommunication.cc +++ b/webrtc/modules/audio_coding/test/TwoWayCommunication.cc @@ -21,13 +21,13 @@ #endif #include "webrtc/common_types.h" -#include "webrtc/engine_configurations.h" #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h" #include "webrtc/modules/audio_coding/test/PCMFile.h" #include "webrtc/modules/audio_coding/test/utility.h" #include "webrtc/system_wrappers/include/trace.h" #include "webrtc/test/gtest.h" #include "webrtc/test/testsupport/fileutils.h" +#include "webrtc/voice_engine_configurations.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/test/delay_test.cc b/webrtc/modules/audio_coding/test/delay_test.cc index 4f9ed533f6..99ec0b82ea 100644 --- a/webrtc/modules/audio_coding/test/delay_test.cc +++ b/webrtc/modules/audio_coding/test/delay_test.cc @@ -16,7 +16,6 @@ #include "gflags/gflags.h" #include "webrtc/common_types.h" -#include "webrtc/engine_configurations.h" #include "webrtc/modules/audio_coding/acm2/acm_common_defs.h" #include "webrtc/modules/audio_coding/include/audio_coding_module.h" #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h" @@ -26,6 +25,7 @@ #include "webrtc/system_wrappers/include/event_wrapper.h" #include "webrtc/test/gtest.h" #include "webrtc/test/testsupport/fileutils.h" +#include "webrtc/voice_engine_configurations.h" DEFINE_string(codec, "isac", "Codec Name"); DEFINE_int32(sample_rate_hz, 16000, "Sampling rate in Hertz."); diff --git a/webrtc/modules/audio_coding/test/opus_test.cc b/webrtc/modules/audio_coding/test/opus_test.cc index d3f7b7aafa..4b3cb6ba5a 100644 --- a/webrtc/modules/audio_coding/test/opus_test.cc +++ b/webrtc/modules/audio_coding/test/opus_test.cc @@ -15,7 +15,6 @@ #include #include "webrtc/common_types.h" -#include "webrtc/engine_configurations.h" #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h" #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h" #include "webrtc/modules/audio_coding/test/TestStereo.h" @@ -23,6 +22,7 @@ #include "webrtc/system_wrappers/include/trace.h" #include "webrtc/test/gtest.h" #include "webrtc/test/testsupport/fileutils.h" +#include "webrtc/voice_engine_configurations.h" namespace webrtc { diff --git a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h index e726c08f75..45d6d2ff6a 100644 --- a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h +++ b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h @@ -15,11 +15,11 @@ #include #include -#include "webrtc/engine_configurations.h" #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h" #include "webrtc/modules/audio_conference_mixer/source/memory_pool.h" #include "webrtc/modules/audio_conference_mixer/source/time_scheduler.h" #include "webrtc/modules/include/module_common_types.h" +#include "webrtc/voice_engine_configurations.h" namespace webrtc { class AudioProcessing; diff --git a/webrtc/modules/audio_mixer/audio_mixer_impl.h b/webrtc/modules/audio_mixer/audio_mixer_impl.h index ecfcbfa621..942196afd2 100644 --- a/webrtc/modules/audio_mixer/audio_mixer_impl.h +++ b/webrtc/modules/audio_mixer/audio_mixer_impl.h @@ -17,13 +17,13 @@ #include "webrtc/base/thread_annotations.h" #include "webrtc/base/thread_checker.h" -#include "webrtc/engine_configurations.h" #include "webrtc/modules/audio_mixer/audio_mixer.h" #include "webrtc/modules/audio_mixer/audio_mixer_defines.h" #include "webrtc/modules/audio_processing/include/audio_processing.h" #include "webrtc/modules/include/module_common_types.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/voice_engine/level_indicator.h" +#include "webrtc/voice_engine_configurations.h" namespace webrtc { diff --git a/webrtc/modules/media_file/media_file_defines.h b/webrtc/modules/media_file/media_file_defines.h index a021a148a5..aa024ce0d6 100644 --- a/webrtc/modules/media_file/media_file_defines.h +++ b/webrtc/modules/media_file/media_file_defines.h @@ -11,9 +11,9 @@ #ifndef WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_DEFINES_H_ #define WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_DEFINES_H_ -#include "webrtc/engine_configurations.h" #include "webrtc/modules/include/module_common_types.h" #include "webrtc/typedefs.h" +#include "webrtc/voice_engine_configurations.h" namespace webrtc { // Callback class for the MediaFile class. diff --git a/webrtc/modules/media_file/media_file_utility.cc b/webrtc/modules/media_file/media_file_utility.cc index 8e7c8f6a02..0096ac59cd 100644 --- a/webrtc/modules/media_file/media_file_utility.cc +++ b/webrtc/modules/media_file/media_file_utility.cc @@ -18,10 +18,10 @@ #include "webrtc/base/format_macros.h" #include "webrtc/common_audio/wav_header.h" #include "webrtc/common_types.h" -#include "webrtc/engine_configurations.h" #include "webrtc/modules/include/module_common_types.h" #include "webrtc/system_wrappers/include/file_wrapper.h" #include "webrtc/system_wrappers/include/trace.h" +#include "webrtc/voice_engine_configurations.h" namespace { diff --git a/webrtc/modules/utility/include/file_player.h b/webrtc/modules/utility/include/file_player.h index 0020514ab3..cafd3a6405 100644 --- a/webrtc/modules/utility/include/file_player.h +++ b/webrtc/modules/utility/include/file_player.h @@ -14,9 +14,9 @@ #include #include "webrtc/common_types.h" -#include "webrtc/engine_configurations.h" #include "webrtc/modules/include/module_common_types.h" #include "webrtc/typedefs.h" +#include "webrtc/voice_engine_configurations.h" namespace webrtc { diff --git a/webrtc/modules/utility/include/file_recorder.h b/webrtc/modules/utility/include/file_recorder.h index 36c755e2da..96bc5cfce6 100644 --- a/webrtc/modules/utility/include/file_recorder.h +++ b/webrtc/modules/utility/include/file_recorder.h @@ -14,10 +14,10 @@ #include #include "webrtc/common_types.h" -#include "webrtc/engine_configurations.h" #include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/media_file/media_file_defines.h" #include "webrtc/typedefs.h" +#include "webrtc/voice_engine_configurations.h" namespace webrtc { diff --git a/webrtc/modules/utility/source/file_player.cc b/webrtc/modules/utility/source/file_player.cc index bf6d26939f..1559e9d91b 100644 --- a/webrtc/modules/utility/source/file_player.cc +++ b/webrtc/modules/utility/source/file_player.cc @@ -12,13 +12,13 @@ #include "webrtc/common_audio/resampler/include/resampler.h" #include "webrtc/common_types.h" -#include "webrtc/engine_configurations.h" #include "webrtc/modules/media_file/media_file.h" #include "webrtc/modules/media_file/media_file_defines.h" #include "webrtc/modules/utility/source/coder.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/system_wrappers/include/logging.h" #include "webrtc/typedefs.h" +#include "webrtc/voice_engine_configurations.h" namespace webrtc { diff --git a/webrtc/modules/utility/source/file_recorder.cc b/webrtc/modules/utility/source/file_recorder.cc index c28c2d11b5..6432a92311 100644 --- a/webrtc/modules/utility/source/file_recorder.cc +++ b/webrtc/modules/utility/source/file_recorder.cc @@ -15,7 +15,6 @@ #include "webrtc/base/platform_thread.h" #include "webrtc/common_audio/resampler/include/resampler.h" #include "webrtc/common_types.h" -#include "webrtc/engine_configurations.h" #include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/media_file/media_file.h" #include "webrtc/modules/media_file/media_file_defines.h" @@ -23,6 +22,7 @@ #include "webrtc/system_wrappers/include/event_wrapper.h" #include "webrtc/system_wrappers/include/logging.h" #include "webrtc/typedefs.h" +#include "webrtc/voice_engine_configurations.h" namespace webrtc { diff --git a/webrtc/modules/video_coding/codec_database.cc b/webrtc/modules/video_coding/codec_database.cc index c8c0b9cfdd..72a1ec322f 100644 --- a/webrtc/modules/video_coding/codec_database.cc +++ b/webrtc/modules/video_coding/codec_database.cc @@ -14,7 +14,6 @@ #include "webrtc/base/checks.h" #include "webrtc/base/logging.h" -#include "webrtc/engine_configurations.h" #include "webrtc/modules/video_coding/codecs/h264/include/h264.h" #include "webrtc/modules/video_coding/codecs/i420/include/i420.h" #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h" diff --git a/webrtc/modules/video_coding/generic_encoder.cc b/webrtc/modules/video_coding/generic_encoder.cc index 1e8362ef09..acef556b24 100644 --- a/webrtc/modules/video_coding/generic_encoder.cc +++ b/webrtc/modules/video_coding/generic_encoder.cc @@ -15,7 +15,6 @@ #include "webrtc/base/checks.h" #include "webrtc/base/logging.h" #include "webrtc/base/trace_event.h" -#include "webrtc/engine_configurations.h" #include "webrtc/modules/video_coding/encoded_frame.h" #include "webrtc/modules/video_coding/media_optimization.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" diff --git a/webrtc/video/rtp_stream_receiver.h b/webrtc/video/rtp_stream_receiver.h index 2af15b0ffc..7069124ba0 100644 --- a/webrtc/video/rtp_stream_receiver.h +++ b/webrtc/video/rtp_stream_receiver.h @@ -18,7 +18,6 @@ #include "webrtc/base/constructormagic.h" #include "webrtc/base/criticalsection.h" -#include "webrtc/engine_configurations.h" #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" diff --git a/webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h b/webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h index 7280a3a343..cec27cb37d 100644 --- a/webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h +++ b/webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h @@ -12,9 +12,9 @@ #define SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_H_ #include "webrtc/common_types.h" -#include "webrtc/engine_configurations.h" #include "webrtc/test/gmock.h" #include "webrtc/test/gtest.h" +#include "webrtc/voice_engine_configurations.h" #include "webrtc/voice_engine/include/voe_audio_processing.h" #include "webrtc/voice_engine/include/voe_base.h" #include "webrtc/voice_engine/include/voe_codec.h" diff --git a/webrtc/voice_engine/test/auto_test/voe_standard_test.cc b/webrtc/voice_engine/test/auto_test/voe_standard_test.cc index 2af6b5bd86..a819bf4812 100644 --- a/webrtc/voice_engine/test/auto_test/voe_standard_test.cc +++ b/webrtc/voice_engine/test/auto_test/voe_standard_test.cc @@ -14,7 +14,6 @@ #include #include -#include "webrtc/engine_configurations.h" #include "webrtc/system_wrappers/include/event_wrapper.h" #include "webrtc/voice_engine/include/voe_neteq_stats.h" #include "webrtc/voice_engine/test/auto_test/automated_mode.h" @@ -22,6 +21,7 @@ #include "webrtc/voice_engine/test/auto_test/voe_stress_test.h" #include "webrtc/voice_engine/test/auto_test/voe_test_defines.h" #include "webrtc/voice_engine/voice_engine_defines.h" +#include "webrtc/voice_engine_configurations.h" DEFINE_bool(include_timing_dependent_tests, true, "If true, we will include tests / parts of tests that are known " diff --git a/webrtc/voice_engine/test/auto_test/voe_test_common.h b/webrtc/voice_engine/test/auto_test/voe_test_common.h index 55787adada..74710ccebe 100644 --- a/webrtc/voice_engine/test/auto_test/voe_test_common.h +++ b/webrtc/voice_engine/test/auto_test/voe_test_common.h @@ -26,7 +26,7 @@ #endif // Read WEBRTC_VOICE_ENGINE_XXX_API compiler flags -#include "webrtc/engine_configurations.h" +#include "webrtc/voice_engine_configurations.h" // Time in ms to test each packet size for each codec #define CODEC_TEST_TIME 400 diff --git a/webrtc/voice_engine/test/auto_test/voe_test_defines.h b/webrtc/voice_engine/test/auto_test/voe_test_defines.h index 93f0387c81..fe0db2c829 100644 --- a/webrtc/voice_engine/test/auto_test/voe_test_defines.h +++ b/webrtc/voice_engine/test/auto_test/voe_test_defines.h @@ -14,11 +14,11 @@ #include "webrtc/voice_engine/test/auto_test/voe_test_common.h" // Read WEBRTC_VOICE_ENGINE_XXX_API compiler flags -#include "webrtc/engine_configurations.h" +#include "webrtc/voice_engine_configurations.h" // Select the tests to execute, list order below is same as they will be // executed. Note that, all settings below will be overriden by sub-API -// settings in engine_configurations.h. +// settings in voice_engine_configurations.h. #define _TEST_BASE_ #define _TEST_RTP_RTCP_ #define _TEST_HARDWARE_ diff --git a/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc b/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc index 350c63684b..97188671bb 100644 --- a/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc +++ b/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc @@ -20,7 +20,6 @@ #include "gflags/gflags.h" #include "webrtc/base/format_macros.h" -#include "webrtc/engine_configurations.h" #include "webrtc/modules/audio_processing/include/audio_processing.h" #include "webrtc/test/gtest.h" #include "webrtc/test/testsupport/fileutils.h" @@ -38,6 +37,7 @@ #include "webrtc/voice_engine/include/voe_video_sync.h" #include "webrtc/voice_engine/include/voe_volume_control.h" #include "webrtc/voice_engine/test/channel_transport/channel_transport.h" +#include "webrtc/voice_engine_configurations.h" DEFINE_bool(use_log_file, false, "Output logs to a file; by default they will be printed to stderr."); diff --git a/webrtc/voice_engine/voice_engine_defines.h b/webrtc/voice_engine/voice_engine_defines.h index f4bdd957e9..9408b5a2bb 100644 --- a/webrtc/voice_engine/voice_engine_defines.h +++ b/webrtc/voice_engine/voice_engine_defines.h @@ -17,8 +17,8 @@ #define WEBRTC_VOICE_ENGINE_VOICE_ENGINE_DEFINES_H #include "webrtc/common_types.h" -#include "webrtc/engine_configurations.h" #include "webrtc/modules/audio_processing/include/audio_processing.h" +#include "webrtc/voice_engine_configurations.h" // ---------------------------------------------------------------------------- // Enumerators diff --git a/webrtc/voice_engine/voice_engine_impl.h b/webrtc/voice_engine/voice_engine_impl.h index 3237001c01..d8aebdc83f 100644 --- a/webrtc/voice_engine/voice_engine_impl.h +++ b/webrtc/voice_engine/voice_engine_impl.h @@ -13,9 +13,9 @@ #include -#include "webrtc/engine_configurations.h" #include "webrtc/system_wrappers/include/atomic32.h" #include "webrtc/voice_engine/voe_base_impl.h" +#include "webrtc/voice_engine_configurations.h" #ifdef WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API #include "webrtc/voice_engine/voe_audio_processing_impl.h" diff --git a/webrtc/engine_configurations.h b/webrtc/voice_engine_configurations.h similarity index 60% rename from webrtc/engine_configurations.h rename to webrtc/voice_engine_configurations.h index 011bdaeff0..a77dc61570 100644 --- a/webrtc/engine_configurations.h +++ b/webrtc/voice_engine_configurations.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_ENGINE_CONFIGURATIONS_H_ -#define WEBRTC_ENGINE_CONFIGURATIONS_H_ +#ifndef WEBRTC_VOICE_ENGINE_CONFIGURATIONS_H_ +#define WEBRTC_VOICE_ENGINE_CONFIGURATIONS_H_ #include "webrtc/typedefs.h" @@ -43,32 +43,4 @@ #define WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API #define WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API -// ============================================================================ -// Platform specific configurations -// ============================================================================ - -// ---------------------------------------------------------------------------- -// VideoEngine Windows -// ---------------------------------------------------------------------------- - -#if defined(_WIN32) -#define DIRECT3D9_RENDERING // Requires DirectX 9. -#endif - -// ---------------------------------------------------------------------------- -// VideoEngine MAC -// ---------------------------------------------------------------------------- - -#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS) -#define COCOA_RENDERING -#endif - -// ---------------------------------------------------------------------------- -// VideoEngine Mobile iPhone -// ---------------------------------------------------------------------------- - -#if defined(WEBRTC_IOS) -#define EAGL_RENDERING -#endif - -#endif // WEBRTC_ENGINE_CONFIGURATIONS_H_ +#endif // WEBRTC_VOICE_ENGINE_CONFIGURATIONS_H_