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

@@ -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) {

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",

webrtc/common.gyp

@@ -16,8 +16,8 @@
         'common_types.h',
         'config.h',
         'config.cc',
-        'engine_configurations.h',
         'typedefs.h',
+        'voice_engine_configurations.h',
       ],
     },
   ],

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 {
 

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.

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 {
 

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 {

webrtc/modules/audio_coding/neteq/audio_decoder_impl.h

@@ -13,7 +13,6 @@
 
 #include <assert.h>
 
-#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 {
 

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 {
 

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 {
 

webrtc/modules/audio_coding/test/TestAllCodecs.cc

@@ -15,7 +15,6 @@
 #include <string>
 
 #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

webrtc/modules/audio_coding/test/TestRedFec.cc

@@ -13,11 +13,11 @@
 #include <assert.h>
 
 #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

webrtc/modules/audio_coding/test/TestStereo.cc

@@ -15,12 +15,12 @@
 #include <string>
 
 #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 {
 

webrtc/modules/audio_coding/test/TestVADDTX.cc

@@ -12,10 +12,10 @@
 
 #include <string>
 
-#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 {
 

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 {
 

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.");

webrtc/modules/audio_coding/test/opus_test.cc

@@ -15,7 +15,6 @@
 #include <string>
 
 #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 {
 

webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h

@@ -15,11 +15,11 @@
 #include <map>
 #include <memory>
 
-#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;

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 {
 

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.

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 {
 

webrtc/modules/utility/include/file_player.h

@@ -14,9 +14,9 @@
 #include <memory>
 
 #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 {
 

webrtc/modules/utility/include/file_recorder.h

@@ -14,10 +14,10 @@
 #include <memory>
 
 #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 {
 

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 {
 

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 {
 

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"

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"

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"

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"

webrtc/voice_engine/test/auto_test/voe_standard_test.cc

@@ -14,7 +14,6 @@
 #include <stdio.h>
 #include <string.h>
 
-#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 "

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

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_

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.");

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

webrtc/voice_engine/voice_engine_impl.h

@@ -13,9 +13,9 @@
 
 #include <memory>
 
-#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"

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_
+#ifndef WEBRTC_VOICE_ENGINE_CONFIGURATIONS_H_
-#define WEBRTC_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
 
-// ============================================================================
+#endif  // WEBRTC_VOICE_ENGINE_CONFIGURATIONS_H_
-//                       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_