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}
This commit is contained in:
mflodman
2016-10-07 07:07:28 +02:00
parent 0b4b72797e
commit 7056be937f
36 changed files with 36 additions and 67 deletions

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@ -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) {

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@ -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",

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

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@ -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 {

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@ -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.

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@ -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 {

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@ -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 {

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@ -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 {

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@ -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 {

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@ -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 {

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@ -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

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@ -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

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@ -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 {

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@ -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 {

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@ -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 {

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@ -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.");

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@ -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 {

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@ -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;

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@ -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 {

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@ -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.

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@ -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 {

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@ -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 {

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@ -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 {

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@ -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 {

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@ -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 {

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@ -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"

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@ -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"

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@ -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"

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@ -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"

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@ -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 "

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@ -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

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@ -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_

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@ -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.");

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@ -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

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@ -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"

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@ -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_