Delete voice_engine_configurations.h

The file was aldready pruned down to the point where it only included
webrtc/typedefs.h. Therefore, all includes of
voice_engine_configurations.h are replaced with typedefs.h, except on
two occasions where it was obvously not needed.

BUG=webrtc:6506

Review-Url: https://codereview.webrtc.org/2553583002
Cr-Commit-Position: refs/heads/master@{#15547}

webrtc/BUILD.gn

@@ -334,7 +334,6 @@ rtc_static_library("webrtc_common") {
     "config.cc",
     "config.h",
     "typedefs.h",
-    "voice_engine_configurations.h",
   ]
 
   if (!build_with_chromium && is_clang) {

webrtc/modules/audio_coding/acm2/acm_codec_database.h

@@ -19,7 +19,7 @@
 #include "webrtc/common_types.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"
+#include "webrtc/typedefs.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/voice_engine_configurations.h"
+#include "webrtc/typedefs.h"
 
 // Checks for enabled codecs, we prevent enabling codecs which are not
 // compatible.

webrtc/modules/audio_coding/acm2/acm_receiver.h

@@ -27,7 +27,6 @@
 #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

@@ -14,7 +14,7 @@
 #include "webrtc/base/format_macros.h"
 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
 #include "webrtc/system_wrappers/include/trace.h"
-#include "webrtc/voice_engine_configurations.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 namespace acm2 {

webrtc/modules/audio_coding/neteq/audio_decoder_impl.h

@@ -20,7 +20,6 @@
 #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

@@ -29,7 +29,7 @@
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/voice_engine_configurations.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 

webrtc/modules/audio_coding/test/RTPFile.cc

@@ -23,7 +23,7 @@
 #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"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 

webrtc/modules/audio_coding/test/TestAllCodecs.cc

@@ -23,7 +23,6 @@
 #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

@@ -18,7 +18,7 @@
 #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"
+#include "webrtc/typedefs.h"
 
 #ifdef SUPPORT_RED_WB
 #undef SUPPORT_RED_WB

webrtc/modules/audio_coding/test/TestStereo.cc

@@ -21,7 +21,7 @@
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/voice_engine_configurations.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 

webrtc/modules/audio_coding/test/TestVADDTX.cc

@@ -16,7 +16,7 @@
 #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"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 

webrtc/modules/audio_coding/test/TwoWayCommunication.cc

@@ -28,7 +28,7 @@
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/voice_engine_configurations.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 

webrtc/modules/audio_coding/test/delay_test.cc

@@ -26,7 +26,7 @@
 #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"
+#include "webrtc/typedefs.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

@@ -23,7 +23,7 @@
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/voice_engine_configurations.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 

webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h

@@ -19,7 +19,7 @@
 #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"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 class AudioProcessing;

webrtc/modules/audio_mixer/audio_mixer_impl.h

@@ -22,7 +22,7 @@
 #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_configurations.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 

webrtc/modules/media_file/media_file_defines.h

@@ -13,7 +13,6 @@
 
 #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

@@ -21,7 +21,7 @@
 #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"
+#include "webrtc/typedefs.h"
 
 namespace {
 

webrtc/modules/utility/include/file_player.h

@@ -16,7 +16,6 @@
 #include "webrtc/common_types.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

@@ -17,7 +17,6 @@
 #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

@@ -18,7 +18,6 @@
 #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

@@ -22,7 +22,6 @@
 #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/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h

@@ -14,7 +14,7 @@
 #include "webrtc/common_types.h"
 #include "webrtc/test/gmock.h"
 #include "webrtc/test/gtest.h"
-#include "webrtc/voice_engine_configurations.h"
+#include "webrtc/typedefs.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

@@ -15,13 +15,13 @@
 #include <string.h>
 
 #include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/typedefs.h"
 #include "webrtc/voice_engine/include/voe_neteq_stats.h"
 #include "webrtc/voice_engine/test/auto_test/automated_mode.h"
 #include "webrtc/voice_engine/test/auto_test/voe_cpu_test.h"
 #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

@@ -25,9 +25,6 @@
 #define TEST_LOG_FLUSH fflush(NULL)
 #endif
 
-// Read WEBRTC_VOICE_ENGINE_XXX_API compiler flags
-#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

@@ -13,9 +13,6 @@
 
 #include "webrtc/voice_engine/test/auto_test/voe_test_common.h"
 
-// Read WEBRTC_VOICE_ENGINE_XXX_API compiler flags
-#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 voice_engine_configurations.h.

webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc

@@ -24,6 +24,7 @@
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/trace_to_stderr.h"
+#include "webrtc/typedefs.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"
@@ -37,7 +38,6 @@
 #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

@@ -18,7 +18,7 @@
 
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/voice_engine_configurations.h"
+#include "webrtc/typedefs.h"
 
 // ----------------------------------------------------------------------------
 //  Enumerators

webrtc/voice_engine/voice_engine_impl.h

@@ -14,7 +14,7 @@
 #include <memory>
 
 #include "webrtc/system_wrappers/include/atomic32.h"
-#include "webrtc/voice_engine_configurations.h"
+#include "webrtc/typedefs.h"
 #include "webrtc/voice_engine/voe_base_impl.h"
 #include "webrtc/voice_engine/voe_audio_processing_impl.h"
 #include "webrtc/voice_engine/voe_codec_impl.h"

webrtc/voice_engine_configurations.h

@@ -1,16 +0,0 @@
-/*
- *  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.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_CONFIGURATIONS_H_
-#define WEBRTC_VOICE_ENGINE_CONFIGURATIONS_H_
-
-#include "webrtc/typedefs.h"
-
-#endif  // WEBRTC_VOICE_ENGINE_CONFIGURATIONS_H_