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
This commit is contained in:
pbos@webrtc.org 2013-05-28 08:13:20 +00:00
parent 7fad4b8c9f
commit 15c1c61e2c
14 changed files with 51 additions and 51 deletions

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@ -11,9 +11,9 @@
#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_ #ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_
#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_ #define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_
#include "audio_conference_mixer_defines.h" #include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h"
#include "module.h" #include "webrtc/modules/interface/module.h"
#include "module_common_types.h" #include "webrtc/modules/interface/module_common_types.h"
namespace webrtc { namespace webrtc {
class AudioMixerOutputReceiver; class AudioMixerOutputReceiver;

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@ -11,9 +11,9 @@
#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_ #ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_ #define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
#include "map_wrapper.h" #include "webrtc/modules/interface/module_common_types.h"
#include "module_common_types.h" #include "webrtc/system_wrappers/interface/map_wrapper.h"
#include "typedefs.h" #include "webrtc/typedefs.h"
namespace webrtc { namespace webrtc {
class MixHistory; class MixHistory;

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@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "audio_conference_mixer_defines.h" #include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h"
#include "audio_conference_mixer_impl.h" #include "webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h"
#include "audio_frame_manipulator.h" #include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h"
#include "audio_processing.h" #include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "critical_section_wrapper.h" #include "webrtc/modules/utility/interface/audio_frame_operations.h"
#include "map_wrapper.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "modules/utility/interface/audio_frame_operations.h" #include "webrtc/system_wrappers/interface/map_wrapper.h"
#include "trace.h" #include "webrtc/system_wrappers/interface/trace.h"
namespace webrtc { namespace webrtc {
namespace { namespace {

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@ -11,14 +11,14 @@
#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_ #ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_
#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_ #define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_
#include "audio_conference_mixer.h" #include "webrtc/engine_configurations.h"
#include "engine_configurations.h" #include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h"
#include "level_indicator.h" #include "webrtc/modules/audio_conference_mixer/source/level_indicator.h"
#include "list_wrapper.h" #include "webrtc/modules/audio_conference_mixer/source/memory_pool.h"
#include "memory_pool.h" #include "webrtc/modules/audio_conference_mixer/source/time_scheduler.h"
#include "module_common_types.h" #include "webrtc/modules/interface/module_common_types.h"
#include "scoped_ptr.h" #include "webrtc/system_wrappers/interface/list_wrapper.h"
#include "time_scheduler.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc { namespace webrtc {
class AudioProcessing; class AudioProcessing;

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@ -8,9 +8,9 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "audio_frame_manipulator.h" #include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h"
#include "module_common_types.h" #include "webrtc/modules/interface/module_common_types.h"
#include "typedefs.h" #include "webrtc/typedefs.h"
namespace { namespace {
// Linear ramping over 80 samples. // Linear ramping over 80 samples.

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@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "level_indicator.h" #include "webrtc/modules/audio_conference_mixer/source/level_indicator.h"
namespace webrtc { namespace webrtc {
// Array for adding smothing to level changes (ad-hoc). // Array for adding smothing to level changes (ad-hoc).

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@ -11,7 +11,7 @@
#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_LEVEL_INDICATOR_H_ #ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_LEVEL_INDICATOR_H_
#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_LEVEL_INDICATOR_H_ #define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_LEVEL_INDICATOR_H_
#include "typedefs.h" #include "webrtc/typedefs.h"
namespace webrtc { namespace webrtc {
class LevelIndicator class LevelIndicator

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@ -13,12 +13,12 @@
#include <assert.h> #include <assert.h>
#include "typedefs.h" #include "webrtc/typedefs.h"
#if _WIN32 #if _WIN32
#include "memory_pool_win.h" #include "webrtc/modules/audio_conference_mixer/source/memory_pool_win.h"
#else #else
#include "memory_pool_posix.h" #include "webrtc/modules/audio_conference_mixer/source/memory_pool_posix.h"
#endif #endif
namespace webrtc { namespace webrtc {

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@ -13,9 +13,9 @@
#include <assert.h> #include <assert.h>
#include "critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "list_wrapper.h" #include "webrtc/system_wrappers/interface/list_wrapper.h"
#include "typedefs.h" #include "webrtc/typedefs.h"
namespace webrtc { namespace webrtc {
template<class MemoryType> template<class MemoryType>

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@ -14,9 +14,9 @@
#include <assert.h> #include <assert.h>
#include <windows.h> #include <windows.h>
#include "aligned_malloc.h" #include "webrtc/system_wrappers/interface/aligned_malloc.h"
#include "atomic32.h" #include "webrtc/system_wrappers/interface/atomic32.h"
#include "typedefs.h" #include "webrtc/typedefs.h"
namespace webrtc { namespace webrtc {
template<class MemoryType> struct MemoryPoolItem; template<class MemoryType> struct MemoryPoolItem;

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@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "critical_section_wrapper.h" #include "webrtc/modules/audio_conference_mixer/source/time_scheduler.h"
#include "time_scheduler.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
namespace webrtc { namespace webrtc {
TimeScheduler::TimeScheduler(const uint32_t periodicityInMs) TimeScheduler::TimeScheduler(const uint32_t periodicityInMs)

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@ -15,7 +15,7 @@
#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_TIME_SCHEDULER_H_ #ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_TIME_SCHEDULER_H_
#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_TIME_SCHEDULER_H_ #define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_TIME_SCHEDULER_H_
#include "tick_util.h" #include "webrtc/system_wrappers/interface/tick_util.h"
namespace webrtc { namespace webrtc {
class CriticalSectionWrapper; class CriticalSectionWrapper;

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@ -12,11 +12,11 @@
#include <iostream> #include <iostream>
#include <time.h> #include <time.h>
#include "functionTest.h" #include "webrtc/common_audio/vad/include/webrtc_vad.h"
#include "event_wrapper.h" #include "webrtc/modules/audio_conference_mixer/test/FunctionTest/functionTest.h"
#include "trace.h" #include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "thread_wrapper.h" #include "webrtc/system_wrappers/interface/thread_wrapper.h"
#include "webrtc_vad.h" #include "webrtc/system_wrappers/interface/trace.h"
#if (defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)) #if (defined(WEBRTC_LINUX) || defined(WEBRTC_MAC))
#include <sys/stat.h> #include <sys/stat.h>

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@ -11,13 +11,13 @@
#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_TEST_FUNCTIONTEST_FUNCTIONTEST_H_ #ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_TEST_FUNCTIONTEST_FUNCTIONTEST_H_
#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_TEST_FUNCTIONTEST_FUNCTIONTEST_H_ #define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_TEST_FUNCTIONTEST_FUNCTIONTEST_H_
#include "module_common_types.h" #include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h"
#include "level_indicator.h" #include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h"
#include "list_wrapper.h" #include "webrtc/modules/audio_conference_mixer/source/level_indicator.h"
#include "map_wrapper.h" #include "webrtc/modules/interface/module_common_types.h"
#include "audio_conference_mixer.h" #include "webrtc/system_wrappers/interface/list_wrapper.h"
#include "audio_conference_mixer_defines.h" #include "webrtc/system_wrappers/interface/map_wrapper.h"
#include "tick_util.h" #include "webrtc/system_wrappers/interface/tick_util.h"
namespace webrtc { namespace webrtc {
class EventWrapper; class EventWrapper;