Include files from webrtc/.. paths in audio_conference_mixer/
BUG=1662 R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1565004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4117 4adac7df-926f-26a2-2b94-8c16560cd09d
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#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_
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#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_
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#include "audio_conference_mixer_defines.h"
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#include "module.h"
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#include "module_common_types.h"
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#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h"
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#include "webrtc/modules/interface/module.h"
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#include "webrtc/modules/interface/module_common_types.h"
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namespace webrtc {
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class AudioMixerOutputReceiver;
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#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
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#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
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#include "map_wrapper.h"
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#include "module_common_types.h"
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#include "typedefs.h"
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#include "webrtc/modules/interface/module_common_types.h"
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#include "webrtc/system_wrappers/interface/map_wrapper.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class MixHistory;
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "audio_conference_mixer_defines.h"
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#include "audio_conference_mixer_impl.h"
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#include "audio_frame_manipulator.h"
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#include "audio_processing.h"
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#include "critical_section_wrapper.h"
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#include "map_wrapper.h"
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#include "modules/utility/interface/audio_frame_operations.h"
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#include "trace.h"
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#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h"
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#include "webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h"
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#include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/modules/utility/interface/audio_frame_operations.h"
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#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
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#include "webrtc/system_wrappers/interface/map_wrapper.h"
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#include "webrtc/system_wrappers/interface/trace.h"
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namespace webrtc {
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namespace {
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#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_
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#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_
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#include "audio_conference_mixer.h"
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#include "engine_configurations.h"
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#include "level_indicator.h"
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#include "list_wrapper.h"
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#include "memory_pool.h"
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#include "module_common_types.h"
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#include "scoped_ptr.h"
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#include "time_scheduler.h"
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#include "webrtc/engine_configurations.h"
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#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h"
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#include "webrtc/modules/audio_conference_mixer/source/level_indicator.h"
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#include "webrtc/modules/audio_conference_mixer/source/memory_pool.h"
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#include "webrtc/modules/audio_conference_mixer/source/time_scheduler.h"
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#include "webrtc/modules/interface/module_common_types.h"
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#include "webrtc/system_wrappers/interface/list_wrapper.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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namespace webrtc {
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class AudioProcessing;
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "audio_frame_manipulator.h"
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#include "module_common_types.h"
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#include "typedefs.h"
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#include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h"
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#include "webrtc/modules/interface/module_common_types.h"
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#include "webrtc/typedefs.h"
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namespace {
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// Linear ramping over 80 samples.
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "level_indicator.h"
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#include "webrtc/modules/audio_conference_mixer/source/level_indicator.h"
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namespace webrtc {
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// Array for adding smothing to level changes (ad-hoc).
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#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_LEVEL_INDICATOR_H_
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#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_LEVEL_INDICATOR_H_
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#include "typedefs.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class LevelIndicator
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#include <assert.h>
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#include "typedefs.h"
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#include "webrtc/typedefs.h"
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#if _WIN32
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#include "memory_pool_win.h"
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#include "webrtc/modules/audio_conference_mixer/source/memory_pool_win.h"
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#else
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#include "memory_pool_posix.h"
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#include "webrtc/modules/audio_conference_mixer/source/memory_pool_posix.h"
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#endif
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namespace webrtc {
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#include <assert.h>
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#include "critical_section_wrapper.h"
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#include "list_wrapper.h"
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#include "typedefs.h"
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#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
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#include "webrtc/system_wrappers/interface/list_wrapper.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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template<class MemoryType>
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#include <assert.h>
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#include <windows.h>
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#include "aligned_malloc.h"
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#include "atomic32.h"
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#include "typedefs.h"
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#include "webrtc/system_wrappers/interface/aligned_malloc.h"
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#include "webrtc/system_wrappers/interface/atomic32.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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template<class MemoryType> struct MemoryPoolItem;
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "critical_section_wrapper.h"
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#include "time_scheduler.h"
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#include "webrtc/modules/audio_conference_mixer/source/time_scheduler.h"
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#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
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namespace webrtc {
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TimeScheduler::TimeScheduler(const uint32_t periodicityInMs)
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#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_TIME_SCHEDULER_H_
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#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_TIME_SCHEDULER_H_
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#include "tick_util.h"
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#include "webrtc/system_wrappers/interface/tick_util.h"
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namespace webrtc {
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class CriticalSectionWrapper;
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#include <iostream>
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#include <time.h>
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#include "functionTest.h"
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#include "event_wrapper.h"
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#include "trace.h"
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#include "thread_wrapper.h"
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#include "webrtc_vad.h"
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#include "webrtc/common_audio/vad/include/webrtc_vad.h"
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#include "webrtc/modules/audio_conference_mixer/test/FunctionTest/functionTest.h"
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#include "webrtc/system_wrappers/interface/event_wrapper.h"
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#include "webrtc/system_wrappers/interface/thread_wrapper.h"
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#include "webrtc/system_wrappers/interface/trace.h"
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#if (defined(WEBRTC_LINUX) || defined(WEBRTC_MAC))
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#include <sys/stat.h>
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#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_TEST_FUNCTIONTEST_FUNCTIONTEST_H_
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#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_TEST_FUNCTIONTEST_FUNCTIONTEST_H_
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#include "module_common_types.h"
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#include "level_indicator.h"
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#include "list_wrapper.h"
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#include "map_wrapper.h"
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#include "audio_conference_mixer.h"
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#include "audio_conference_mixer_defines.h"
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#include "tick_util.h"
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#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h"
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#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h"
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#include "webrtc/modules/audio_conference_mixer/source/level_indicator.h"
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#include "webrtc/modules/interface/module_common_types.h"
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#include "webrtc/system_wrappers/interface/list_wrapper.h"
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#include "webrtc/system_wrappers/interface/map_wrapper.h"
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#include "webrtc/system_wrappers/interface/tick_util.h"
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namespace webrtc {
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class EventWrapper;
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