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Delete unneeded includes of deprecated system_wrappers include files.

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Deletes left-over includes of trace.h and critical_section_wrapper.h.

BUG=webrtc:7035

Review-Url: https://codereview.webrtc.org/2784873002
Cr-Commit-Position: refs/heads/master@{#17460}
This commit is contained in:
nisse
2017-03-30 00:31:15 -07:00
committed by Commit bot
parent e5ad5ca06a
commit 0ffdcc51bc
38 changed files with 3 additions and 40 deletions
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2
webrtc/call/bitrate_estimator_tests.cc
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@ -17,8 +17,6 @@
#include "webrtc/base/logging.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/call/call.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/test/call_test.h"
#include "webrtc/test/direct_transport.h"
#include "webrtc/test/encoder_settings.h"

1
webrtc/call/call.cc
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@ -48,7 +48,6 @@
#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/cpu_info.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/metrics.h"
#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"

1
webrtc/call/call_perf_tests.cc
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@ -22,7 +22,6 @@
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/metrics_default.h"
#include "webrtc/test/call_test.h"
#include "webrtc/test/direct_transport.h"

1
webrtc/common_video/incoming_video_stream.cc
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@ -15,7 +15,6 @@
#include "webrtc/base/timeutils.h"
#include "webrtc/base/trace_event.h"
#include "webrtc/common_video/video_render_frames.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
namespace webrtc {

1
webrtc/modules/audio_coding/acm2/acm_codec_database.cc
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@ -20,7 +20,6 @@
#include <assert.h>
#include "webrtc/base/checks.h"
#include "webrtc/system_wrappers/include/trace.h"
#if ((defined WEBRTC_CODEC_ISAC) && (defined WEBRTC_CODEC_ISACFX))
#error iSAC and iSACFX codecs cannot be enabled at the same time

1
webrtc/modules/audio_coding/acm2/acm_receiver.cc
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@ -26,7 +26,6 @@
#include "webrtc/modules/audio_coding/acm2/call_statistics.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
namespace webrtc {

1
webrtc/modules/audio_coding/test/TestRedFec.cc
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@ -16,7 +16,6 @@
#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.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/typedefs.h"

1
webrtc/modules/audio_coding/test/TestStereo.cc
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@ -18,7 +18,6 @@
#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.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/typedefs.h"

1
webrtc/modules/audio_coding/test/TwoWayCommunication.cc
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@ -25,7 +25,6 @@
#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.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/typedefs.h"

1
webrtc/modules/audio_coding/test/iSACTest.cc
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@ -26,7 +26,6 @@
#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.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/testsupport/fileutils.h"
namespace webrtc {

1
webrtc/modules/audio_coding/test/opus_test.cc
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@ -20,7 +20,6 @@
#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "webrtc/modules/audio_coding/test/TestStereo.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/typedefs.h"

1
webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
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@ -13,7 +13,6 @@
#include "webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h"
#include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"
namespace webrtc {

1
webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h
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@ -13,7 +13,6 @@
#include "webrtc/modules/audio_device/include/audio_device.h"
#include "webrtc/modules/audio_device/linux/pulseaudiosymboltable_linux.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/typedefs.h"
#include "webrtc/base/thread_checker.h"

2
webrtc/modules/audio_device/linux/latebindingsymboltable_linux.cc
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@ -10,6 +10,8 @@
#include "webrtc/modules/audio_device/linux/latebindingsymboltable_linux.h"
#include "webrtc/system_wrappers/include/trace.h"
#ifdef WEBRTC_LINUX
#include <dlfcn.h>
#endif

3
webrtc/modules/audio_device/linux/latebindingsymboltable_linux.h
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@ -16,7 +16,6 @@
#include <string.h>
#include "webrtc/base/constructormagic.h"
#include "webrtc/system_wrappers/include/trace.h"
// This file provides macros for creating "symbol table" classes to simplify the
// dynamic loading of symbols from DLLs. Currently the implementation only
@ -81,8 +80,6 @@ class LateBindingSymbolTable {
if (undefined_symbols_) {
// We do not attempt to load again because repeated attempts are not
// likely to succeed and DLL loading is costly.
//WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
// "We know there are undefined symbols");
return false;
}
handle_ = InternalLoadDll(kDllName);

1
webrtc/modules/audio_device/test/audio_device_test_defines.h
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@ -13,7 +13,6 @@
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_device/include/audio_device.h"
#include "webrtc/system_wrappers/include/trace.h"
#ifdef _WIN32
#define MACRO_DEFAULT_DEVICE AudioDeviceModule::kDefaultDevice

1
webrtc/modules/audio_mixer/audio_mixer_impl.h
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@ -22,7 +22,6 @@
#include "webrtc/modules/audio_mixer/output_rate_calculator.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/typedefs.h"
namespace webrtc {

1
webrtc/modules/audio_processing/level_estimator_impl.cc
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@ -13,7 +13,6 @@
#include "webrtc/base/array_view.h"
#include "webrtc/modules/audio_processing/audio_buffer.h"
#include "webrtc/modules/audio_processing/rms_level.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
namespace webrtc {

1
webrtc/modules/audio_processing/low_cut_filter.cc
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@ -12,7 +12,6 @@
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_processing/audio_buffer.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
namespace webrtc {
namespace {

1
webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
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@ -18,7 +18,6 @@
#include <vector>
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
namespace webrtc {

1
webrtc/modules/remote_bitrate_estimator/overuse_detector.cc
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@ -23,7 +23,6 @@
#include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/system_wrappers/include/field_trial.h"
#include "webrtc/system_wrappers/include/trace.h"
namespace webrtc {

1
webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
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@ -20,7 +20,6 @@
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/pacing/paced_sender.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/metrics.h"
#include "webrtc/typedefs.h"

1
webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h
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@ -26,7 +26,6 @@
#include "webrtc/modules/remote_bitrate_estimator/inter_arrival.h"
#include "webrtc/modules/remote_bitrate_estimator/overuse_detector.h"
#include "webrtc/modules/remote_bitrate_estimator/overuse_estimator.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
namespace webrtc {

1
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
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@ -19,7 +19,6 @@
#include "webrtc/base/logging.h"
#include "webrtc/common_types.h"
#include "webrtc/config.h"
#include "webrtc/system_wrappers/include/trace.h"
#ifdef _WIN32
// Disable warning C4355: 'this' : used in base member initializer list.

1
webrtc/modules/video_capture/objc/video_capture.mm
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@ -16,7 +16,6 @@
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/modules/video_capture/objc/device_info_objc.h"
#include "webrtc/modules/video_capture/objc/rtc_video_capture_objc.h"
#include "webrtc/system_wrappers/include/trace.h"
using namespace webrtc;
using namespace videocapturemodule;

1
webrtc/modules/video_coding/codecs/tools/video_quality_measurement.cc
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@ -27,7 +27,6 @@
#include "webrtc/modules/video_coding/codecs/test/videoprocessor.h"
#include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
#include "webrtc/modules/video_coding/include/video_coding.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/test/testsupport/frame_reader.h"
#include "webrtc/test/testsupport/frame_writer.h"
#include "webrtc/test/testsupport/metrics/video_metrics.h"

1
webrtc/modules/video_coding/generic_encoder.cc
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@ -18,7 +18,6 @@
#include "webrtc/base/trace_event.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"
namespace webrtc {

1
webrtc/modules/video_coding/media_opt_util.h
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@ -18,7 +18,6 @@
#include "webrtc/base/numerics/exp_filter.h"
#include "webrtc/modules/video_coding/internal_defines.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/typedefs.h"
namespace webrtc {

2
webrtc/modules/video_coding/media_optimization.h
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@ -14,10 +14,10 @@
#include <list>
#include <memory>
#include "webrtc/base/criticalsection.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_coding/include/video_coding.h"
#include "webrtc/modules/video_coding/media_opt_util.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
namespace webrtc {

1
webrtc/modules/video_coding/receiver_unittest.cc
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@ -22,7 +22,6 @@
#include "webrtc/modules/video_coding/test/test_util.h"
#include "webrtc/modules/video_coding/timing.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/test/gtest.h"
namespace webrtc {

1
webrtc/modules/video_coding/timing_unittest.cc
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@ -16,7 +16,6 @@
#include "webrtc/modules/video_coding/internal_defines.h"
#include "webrtc/modules/video_coding/timing.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/testsupport/fileutils.h"

1
webrtc/modules/video_coding/utility/frame_dropper.cc
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@ -13,7 +13,6 @@
#include <algorithm>
#include "webrtc/base/logging.h"
#include "webrtc/system_wrappers/include/trace.h"
namespace webrtc {

1
webrtc/system_wrappers/source/condition_variable_unittest.cc
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@ -16,7 +16,6 @@
#include "webrtc/base/platform_thread.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/system_wrappers/source/condition_variable_event_win.h"
#include "webrtc/test/gtest.h"

1
webrtc/system_wrappers/source/critical_section_unittest.cc
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@ -12,7 +12,6 @@
#include "webrtc/base/platform_thread.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/test/gtest.h"
namespace webrtc {

1
webrtc/video/rtp_stream_receiver.cc
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@ -37,7 +37,6 @@
#include "webrtc/system_wrappers/include/field_trial.h"
#include "webrtc/system_wrappers/include/metrics.h"
#include "webrtc/system_wrappers/include/timestamp_extrapolator.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/video/receive_statistics_proxy.h"
#include "webrtc/video/vie_remb.h"

1
webrtc/video/vie_remb.cc
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@ -16,7 +16,6 @@
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/system_wrappers/include/trace.h"
namespace webrtc {

1
webrtc/voice_engine/file_player.cc
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@ -14,7 +14,6 @@
#include "webrtc/common_types.h"
#include "webrtc/modules/media_file/media_file.h"
#include "webrtc/modules/media_file/media_file_defines.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/coder.h"

1
webrtc/voice_engine/voe_network_impl.cc
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@ -13,7 +13,6 @@
#include "webrtc/base/checks.h"
#include "webrtc/base/format_macros.h"
#include "webrtc/base/logging.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/voice_engine/channel.h"
#include "webrtc/voice_engine/include/voe_errors.h"
#include "webrtc/voice_engine/voice_engine_impl.h"