[RELEASE] Release v1.0.0

.travis.yml

@@ -1,11 +1,7 @@
-language:
-  - cpp
+language: cpp
 
-sudo: false
-
-os:
-  - linux
-  - osx
+sudo: required
+dist: trusty
 
 branches:
   only:
@@ -18,33 +14,81 @@ addons:
       - ubuntu-toolchain-r-test
     packages:
       - g++-4.9
+      - expect
+      - binutils-mingw-w64-x86-64 # 64bit MinGW
+      - gcc-mingw-w64-x86-64
+      - g++-mingw-w64-x86-64
+
+matrix:
+  include:
+    - os: linux
+      env: CONF=release BUILDER=gcc TARGET=Linux TAG=Linux COMPILATOR_OPTION="--compilator-version=4.9" GCOV=--gcov
+      compiler: gcc
+    - os: linux
+      env: CONF=debug BUILDER=clang TARGET=Linux
+      compiler: clang
+    - os: linux
+      env: CONF=release BUILDER=gcc TARGET=Windows TAG=Mingw
+      compiler: x86_64-w64-mingw32-gcc
+    - os: linux
+      env: CONF=release BUILDER=gcc TARGET=Android TAG=Android DISABLE_PACKAGE=-p
+      compiler: gcc
+    - os: osx
+      env: CONF=release BUILDER=clang TARGET=MacOs TAG=MacOs
+      compiler: clang
+    - os: osx
+      env: CONF=release BUILDER=clang TARGET=IOs TAG=IOs
+      compiler: clang
+
 
 install:
+  - cd ..
   - pip install --user lutin
-
-env:
-  - CONF=debug   BOARD=Linux BUILDER=clang GCOV=
-  - CONF=release BOARD=Linux BUILDER=clang GCOV=
-  - CONF=debug   BOARD=Linux BUILDER=gcc   GCOV=
-  - CONF=release BOARD=Linux BUILDER=gcc   GCOV=
-  - CONF=debug   BOARD=Linux BUILDER=gcc   GCOV=--gcov
+  - if [ "$TAG" == "Android" ]; then
+        git clone --depth 1 --branch master https://github.com/HeeroYui/android-download-tool;
+        ./android-download-tool/dl-android.sh;
+    fi
+  - git clone --depth 1 --branch master https://github.com/atria-soft/ci.git
+  - cd -
 
 before_script:
   - cd ..
-  - wget http://atria-soft.com/ci/coverage_send.py
-  - wget http://atria-soft.com/ci/test_send.py
-  - wget http://atria-soft.com/ci/warning_send.py
-  - git clone https://github.com/atria-soft/etk.git
-  - git clone https://github.com/musicdsp/audio.git
+  - git clone https://github.com/atria-soft/etk.git -b $TRAVIS_BRANCH
+  - git clone https://github.com/atria-soft/elog.git -b $TRAVIS_BRANCH
+  - git clone https://github.com/atria-soft/ememory.git -b $TRAVIS_BRANCH
+  - git clone https://github.com/atria-soft/echrono.git -b $TRAVIS_BRANCH
+  - git clone https://github.com/atria-soft/ethread.git -b $TRAVIS_BRANCH
+  - git clone https://github.com/atria-soft/ejson.git -b $TRAVIS_BRANCH
+  - git clone https://github.com/atria-soft/jvm-basics.git -b $TRAVIS_BRANCH
+  - git clone https://github.com/musicdsp/audio.git -b $TRAVIS_BRANCH
+  - git clone https://github.com/generic-library/gtest-lutin.git --recursive
+  - git clone https://github.com/generic-library/z-lutin.git --recursive
   - pwd
   - ls -l
-  - if [ "$BUILDER" == "gcc" ]; then COMPILATOR_OPTION="--compilator-version=4.9"; else COMPILATOR_OPTION=""; fi
+  - if [ "$TRAVIS_OS_NAME" == "osx" ]; then
+        export PATH=$PATH:/Users/travis/Library/Python/2.7/bin/;
+    fi
+  - ./ci/build_send.py --tag=$TAG --status=START;
+
 
 script:
-  - lutin -w -j4 -C -P -c $BUILDER $COMPILATOR_OPTION -m $CONF $GCOV -p audio-orchestra
-#  - ./out/Linux_x86_64/$CONF/staging/$BUILDER/audio-orchestra/usr/bin/audio-orchestra -l6
+  - lutin -w -j4 -C -P -t$TARGET -c $BUILDER $COMPILATOR_OPTION $BUS -m $CONF $GCOV $DISABLE_PACKAGE audio-orchestra; STATUS=$?
+  - ./ci/build_send.py --tag=$TAG --status="$STATUS";
+
+after_script:
+  - if [ "$GCOV" != "" ]; then
+        ./ci/warning_send.py --find-path ./out/Linux_x86_64/$CONF/build/$BUILDER/audio-orchestra/;
+    fi
+  #- lutin -w -j4 -C -P -t$TARGET -c $BUILDER $COMPILATOR_OPTION $BUS -m $CONF $GCOV $DISABLE_PACKAGE audio-orchestra-test?run:--elog-level=3 | tee out_test.txt
+  #- if [ "$GCOV" != "" ]; then
+  #      ./ci/test_send.py --file=out_test.txt;
+  #      lutin -C -P -t $TARGET -c $BUILDER $COMPILATOR_OPTION $BUS -m $CONF -p audio-orchestra?gcov;
+  #      ./ci/coverage_send.py --json=out/Linux_x86_64/$CONF/build/$BUILDER/audio-orchestra/audio-orchestra_coverage.json;
+  #  fi
 
 notifications:
   email:
     - yui.heero@gmail.com
 
+
+

audio/orchestra/Api.cpp

@@ -5,43 +5,40 @@
  * @fork from RTAudio
  */
 
-//#include <etk/types.h>
-#include <audio/orchestra/Interface.h>
-#include <audio/orchestra/debug.h>
-#include <iostream>
-#include <cstdlib>
-#include <cstring>
-#include <climits>
+//#include <etk/types.hpp>
+#include <audio/orchestra/Interface.hpp>
+#include <audio/orchestra/debug.hpp>
+#include <etk/types.hpp>
 
 // Static variable definitions.
-const std::vector<uint32_t>& audio::orchestra::genericSampleRate() {
-	static std::vector<uint32_t> list;
+const etk::Vector<uint32_t>& audio::orchestra::genericSampleRate() {
+	static etk::Vector<uint32_t> list;
 	if (list.size() == 0) {
-		list.push_back(4000);
-		list.push_back(5512);
-		list.push_back(8000);
-		list.push_back(9600);
-		list.push_back(11025);
-		list.push_back(16000);
-		list.push_back(22050);
-		list.push_back(32000);
-		list.push_back(44100);
-		list.push_back(48000);
-		list.push_back(64000);
-		list.push_back(88200);
-		list.push_back(96000);
-		list.push_back(128000);
-		list.push_back(176400);
-		list.push_back(192000);
-		list.push_back(256000);
+		list.pushBack(4000);
+		list.pushBack(5512);
+		list.pushBack(8000);
+		list.pushBack(9600);
+		list.pushBack(11025);
+		list.pushBack(16000);
+		list.pushBack(22050);
+		list.pushBack(32000);
+		list.pushBack(44100);
+		list.pushBack(48000);
+		list.pushBack(64000);
+		list.pushBack(88200);
+		list.pushBack(96000);
+		list.pushBack(128000);
+		list.pushBack(176400);
+		list.pushBack(192000);
+		list.pushBack(256000);
 	}
 	return list;
 };
 
 
 audio::orchestra::Api::Api() :
-  m_callback(nullptr),
-  m_deviceBuffer(nullptr) {
+  m_callback(null),
+  m_deviceBuffer(null) {
 	m_device[0] = 11111;
 	m_device[1] = 11111;
 	m_state = audio::orchestra::state::closed;
@@ -55,34 +52,34 @@ audio::orchestra::Api::~Api() {
 enum audio::orchestra::error audio::orchestra::Api::startStream() {
 	ATA_VERBOSE("Start Stream");
 	m_startTime = audio::Time::now();
-	m_duration = std::chrono::microseconds(0);
+	m_duration = echrono::microseconds(0);
 	return audio::orchestra::error_none;
 }
 
 enum audio::orchestra::error audio::orchestra::Api::openStream(audio::orchestra::StreamParameters* _oParams,
-                                                 audio::orchestra::StreamParameters* _iParams,
-                                                 enum audio::format _format,
-                                                 uint32_t _sampleRate,
-                                                 uint32_t* _bufferFrames,
-                                                 audio::orchestra::AirTAudioCallback _callback,
-                                                 const audio::orchestra::StreamOptions& _options) {
+                                                               audio::orchestra::StreamParameters* _iParams,
+                                                               enum audio::format _format,
+                                                               uint32_t _sampleRate,
+                                                               uint32_t* _bufferFrames,
+                                                               audio::orchestra::AirTAudioCallback _callback,
+                                                               const audio::orchestra::StreamOptions& _options) {
 	if (m_state != audio::orchestra::state::closed) {
 		ATA_ERROR("a stream is already open!");
 		return audio::orchestra::error_invalidUse;
 	}
-	if (    _oParams != nullptr
+	if (    _oParams != null
 	     && _oParams->nChannels < 1) {
-		ATA_ERROR("a non-nullptr output StreamParameters structure cannot have an nChannels value less than one.");
+		ATA_ERROR("a non-null output StreamParameters structure cannot have an nChannels value less than one.");
 		return audio::orchestra::error_invalidUse;
 	}
-	if (    _iParams != nullptr
+	if (    _iParams != null
 	     && _iParams->nChannels < 1) {
-		ATA_ERROR("a non-nullptr input StreamParameters structure cannot have an nChannels value less than one.");
+		ATA_ERROR("a non-null input StreamParameters structure cannot have an nChannels value less than one.");
 		return audio::orchestra::error_invalidUse;
 	}
-	if (    _oParams == nullptr
-	     && _iParams == nullptr) {
-		ATA_ERROR("input and output StreamParameters structures are both nullptr!");
+	if (    _oParams == null
+	     && _iParams == null) {
+		ATA_ERROR("input and output StreamParameters structures are both null!");
 		return audio::orchestra::error_invalidUse;
 	}
 	if (audio::getFormatBytes(_format) == 0) {
@@ -91,7 +88,7 @@ enum audio::orchestra::error audio::orchestra::Api::openStream(audio::orchestra:
 	}
 	uint32_t nDevices = getDeviceCount();
 	uint32_t oChannels = 0;
-	if (_oParams != nullptr) {
+	if (_oParams != null) {
 		oChannels = _oParams->nChannels;
 		if (    _oParams->deviceId >= nDevices
 		     && _oParams->deviceName == "") {
@@ -100,7 +97,7 @@ enum audio::orchestra::error audio::orchestra::Api::openStream(audio::orchestra:
 		}
 	}
 	uint32_t iChannels = 0;
-	if (_iParams != nullptr) {
+	if (_iParams != null) {
 		iChannels = _iParams->nChannels;
 		if (    _iParams->deviceId >= nDevices
 		     && _iParams->deviceName == "") {
@@ -250,8 +247,8 @@ void audio::orchestra::Api::clearStreamInfo() {
 	m_userFormat = audio::format_unknow;
 	m_startTime = audio::Time();
 	m_duration = audio::Duration(0);
-	m_deviceBuffer = nullptr;
-	m_callback = nullptr;
+	m_deviceBuffer = null;
+	m_callback = null;
 	for (int32_t iii=0; iii<2; ++iii) {
 		m_device[iii] = 11111;
 		m_doConvertBuffer[iii] = false;
@@ -295,21 +292,21 @@ void audio::orchestra::Api::setConvertInfo(audio::orchestra::mode _mode, uint32_
 	if (m_deviceInterleaved[idTable] == false) {
 		if (_mode == audio::orchestra::mode_input) {
 			for (int32_t kkk=0; kkk<m_convertInfo[idTable].channels; ++kkk) {
-				m_convertInfo[idTable].inOffset.push_back(kkk * m_bufferSize);
-				m_convertInfo[idTable].outOffset.push_back(kkk);
+				m_convertInfo[idTable].inOffset.pushBack(kkk * m_bufferSize);
+				m_convertInfo[idTable].outOffset.pushBack(kkk);
 				m_convertInfo[idTable].inJump = 1;
 			}
 		} else {
 			for (int32_t kkk=0; kkk<m_convertInfo[idTable].channels; ++kkk) {
-				m_convertInfo[idTable].inOffset.push_back(kkk);
-				m_convertInfo[idTable].outOffset.push_back(kkk * m_bufferSize);
+				m_convertInfo[idTable].inOffset.pushBack(kkk);
+				m_convertInfo[idTable].outOffset.pushBack(kkk * m_bufferSize);
 				m_convertInfo[idTable].outJump = 1;
 			}
 		}
 	} else { // no (de)interleaving
 		for (int32_t kkk=0; kkk<m_convertInfo[idTable].channels; ++kkk) {
-			m_convertInfo[idTable].inOffset.push_back(kkk);
-			m_convertInfo[idTable].outOffset.push_back(kkk);
+			m_convertInfo[idTable].inOffset.pushBack(kkk);
+			m_convertInfo[idTable].outOffset.pushBack(kkk);
 		}
 	}
 

audio/orchestra/Api.hpp

@@ -6,33 +6,38 @@
  */
 #pragma once
 
-#include <sstream>
-#include <audio/orchestra/debug.h>
-#include <audio/orchestra/type.h>
-#include <audio/orchestra/state.h>
-#include <audio/orchestra/mode.h>
-#include <audio/Time.h>
-#include <audio/Duration.h>
-#include <ememory/memory.h>
-
+#include <etk/Stream.hpp>
+#include <audio/orchestra/debug.hpp>
+#include <audio/orchestra/type.hpp>
+#include <audio/orchestra/state.hpp>
+#include <audio/orchestra/mode.hpp>
+#include <audio/Time.hpp>
+#include <audio/Duration.hpp>
+#include <ememory/memory.hpp>
+/**
+ * @brief Audio library namespace
+ */
 namespace audio {
+	/**
+	 * @brief Audio-orchestra library namespace
+	 */
 	namespace orchestra {
-		const std::vector<uint32_t>& genericSampleRate();
+		const etk::Vector<uint32_t>& genericSampleRate();
 		/**
 		 * @brief airtaudio callback function prototype.
-		 * @param _inputBuffer For input (or duplex) streams, this buffer will hold _nbChunk of input audio chunk (nullptr if no data).
+		 * @param _inputBuffer For input (or duplex) streams, this buffer will hold _nbChunk of input audio chunk (null if no data).
 		 * @param _timeInput Timestamp of the first buffer sample (recording time).
-		 * @param _outputBuffer For output (or duplex) streams, the client should write _nbChunk of audio chunk into this buffer (nullptr if no data).
+		 * @param _outputBuffer For output (or duplex) streams, the client should write _nbChunk of audio chunk into this buffer (null if no data).
 		 * @param _timeOutput Timestamp of the first buffer sample (playing time).
 		 * @param _nbChunk The number of chunk of input or output chunk in the buffer (same size).
 		 * @param _status List of error that occured in the laps of time.
 		 */
-		typedef std::function<int32_t (const void* _inputBuffer,
-		                                 const audio::Time& _timeInput,
-		                                 void* _outputBuffer,
-		                                 const audio::Time& _timeOutput,
-		                                 uint32_t _nbChunk,
-		                                 const std::vector<audio::orchestra::status>& _status)> AirTAudioCallback;
+		typedef etk::Function<int32_t (const void* _inputBuffer,
+		                               const audio::Time& _timeInput,
+		                               void* _outputBuffer,
+		                               const audio::Time& _timeOutput,
+		                               uint32_t _nbChunk,
+		                               const etk::Vector<audio::orchestra::status>& _status)> AirTAudioCallback;
 		// A protected structure used for buffer conversion.
 		class ConvertInfo {
 			public:
@@ -41,24 +46,24 @@ namespace audio {
 				int32_t outJump;
 				enum audio::format inFormat;
 				enum audio::format outFormat;
-				std::vector<int> inOffset;
-				std::vector<int> outOffset;
+				etk::Vector<int> inOffset;
+				etk::Vector<int> outOffset;
 		};
 	
 		class Api : public ememory::EnableSharedFromThis<Api>{
 			protected:
-				std::string m_name;
+				etk::String m_name;
 			public:
 				Api();
 				virtual ~Api();
-				void setName(const std::string& _name) {
+				void setName(const etk::String& _name) {
 					m_name = _name;
 				}
-				virtual const std::string& getCurrentApi() = 0;
+				virtual const etk::String& getCurrentApi() = 0;
 				virtual uint32_t getDeviceCount() = 0;
 				virtual audio::orchestra::DeviceInfo getDeviceInfo(uint32_t _device) = 0;
 				// TODO : Check API ...
-				virtual	bool getNamedDeviceInfo(const std::string& _deviceName, audio::orchestra::DeviceInfo& _info) {
+				virtual	bool getNamedDeviceInfo(const etk::String& _deviceName, audio::orchestra::DeviceInfo& _info) {
 					return false;
 				}
 				virtual uint32_t getDefaultInputDevice();
@@ -85,12 +90,12 @@ namespace audio {
 				}
 				
 			protected:
-				mutable std::mutex m_mutex;
+				mutable ethread::Mutex m_mutex;
 				audio::orchestra::AirTAudioCallback m_callback;
 				uint32_t m_device[2]; // Playback and record, respectively.
 				enum audio::orchestra::mode m_mode; // audio::orchestra::mode_output, audio::orchestra::mode_input, or audio::orchestra::mode_duplex.
 				enum audio::orchestra::state m_state; // STOPPED, RUNNING, or CLOSED
-				std::vector<char> m_userBuffer[2]; // Playback and record, respectively.
+				etk::Vector<char> m_userBuffer[2]; // Playback and record, respectively.
 				char *m_deviceBuffer;
 				bool m_doConvertBuffer[2]; // Playback and record, respectively.
 				bool m_deviceInterleaved[2]; // Playback and record, respectively.
@@ -125,7 +130,7 @@ namespace audio {
 				                  enum audio::format _format,
 				                  uint32_t *_bufferSize,
 				                  const audio::orchestra::StreamOptions& _options);
-				virtual bool openName(const std::string& _deviceName,
+				virtual bool openName(const etk::String& _deviceName,
 				                      audio::orchestra::mode _mode,
 				                      uint32_t _channels,
 				                      uint32_t _firstChannel,

audio/orchestra/DeviceInfo.cpp

@@ -5,14 +5,13 @@
  * @fork from RTAudio
  */
 
-//#include <etk/types.h>
-#include <audio/orchestra/debug.h>
-#include <audio/orchestra/DeviceInfo.h>
-#include <etk/stdTools.h>
-#include <iostream>
+//#include <etk/types.hpp>
+#include <audio/orchestra/debug.hpp>
+#include <audio/orchestra/DeviceInfo.hpp>
+#include <etk/stdTools.hpp>
 
 void audio::orchestra::DeviceInfo::display(int32_t _tabNumber) const {
-	std::string space;
+	etk::String space;
 	for (int32_t iii=0; iii<_tabNumber; ++iii) {
 		space += "    ";
 	}
@@ -42,7 +41,7 @@ void audio::orchestra::DeviceInfo::clear() {
 	isDefault = false;
 }
 
-std::ostream& audio::orchestra::operator <<(std::ostream& _os, const audio::orchestra::DeviceInfo& _obj) {
+etk::Stream& audio::orchestra::operator <<(etk::Stream& _os, const audio::orchestra::DeviceInfo& _obj) {
 	_os << "{";
 	if (_obj.isCorrect == false) {
 		_os << "NOT CORRECT INFORAMATIONS";

audio/orchestra/DeviceInfo.hpp

@@ -6,8 +6,8 @@
  */
 #pragma once
 
-#include <audio/format.h>
-#include <audio/channel.h>
+#include <audio/format.hpp>
+#include <audio/channel.hpp>
 
 namespace audio {
 	namespace orchestra {
@@ -18,11 +18,11 @@ namespace audio {
 			public:
 				bool isCorrect; //!< the information is correct (the system can return information incorect).
 				bool input; //!< true if the device in an input; false: output.
-				std::string name; //!< Character string device identifier.
-				std::string desc; //!< description of the device
-				std::vector<audio::channel> channels; //!< Channels interfaces.
-				std::vector<uint32_t> sampleRates; //!< Supported sample rates (queried from list of standard rates).
-				std::vector<audio::format> nativeFormats; //!< Bit mask of supported data formats.
+				etk::String name; //!< Character string device identifier.
+				etk::String desc; //!< description of the device
+				etk::Vector<audio::channel> channels; //!< Channels interfaces.
+				etk::Vector<uint32_t> sampleRates; //!< Supported sample rates (queried from list of standard rates).
+				etk::Vector<audio::format> nativeFormats; //!< Bit mask of supported data formats.
 				bool isDefault; //! is default input/output
 				// Default constructor.
 				DeviceInfo() :
@@ -43,7 +43,7 @@ namespace audio {
 				 */
 				void clear();
 		};
-		std::ostream& operator <<(std::ostream& _os, const audio::orchestra::DeviceInfo& _obj);
+		etk::Stream& operator <<(etk::Stream& _os, const audio::orchestra::DeviceInfo& _obj);
 	}
 }
 

audio/orchestra/Flags.cpp

@@ -5,5 +5,5 @@
  * @fork from RTAudio
  */
 
-#include <audio/orchestra/Flags.h>
-#include <audio/orchestra/debug.h>
+#include <audio/orchestra/Flags.hpp>
+#include <audio/orchestra/debug.hpp>

audio/orchestra/Flags.hpp

@@ -6,7 +6,7 @@
  */
 #pragma once
 
-#include <etk/types.h>
+#include <etk/types.hpp>
 
 namespace audio {
 	namespace orchestra {

audio/orchestra/Interface.cpp

@@ -5,40 +5,39 @@
  * @fork from RTAudio
  */
 
-//#include <etk/types.h>
-#include <audio/orchestra/Interface.h>
-#include <audio/orchestra/debug.h>
-#include <iostream>
-#include <audio/orchestra/api/Alsa.h>
-#include <audio/orchestra/api/Android.h>
-#include <audio/orchestra/api/Asio.h>
-#include <audio/orchestra/api/Core.h>
-#include <audio/orchestra/api/CoreIos.h>
-#include <audio/orchestra/api/Ds.h>
-#include <audio/orchestra/api/Dummy.h>
-#include <audio/orchestra/api/Jack.h>
-#include <audio/orchestra/api/Pulse.h>
+//#include <etk/types.hpp>
+#include <audio/orchestra/Interface.hpp>
+#include <audio/orchestra/debug.hpp>
+#include <audio/orchestra/api/Alsa.hpp>
+#include <audio/orchestra/api/Android.hpp>
+#include <audio/orchestra/api/Asio.hpp>
+#include <audio/orchestra/api/Core.hpp>
+#include <audio/orchestra/api/CoreIos.hpp>
+#include <audio/orchestra/api/Ds.hpp>
+#include <audio/orchestra/api/Dummy.hpp>
+#include <audio/orchestra/api/Jack.hpp>
+#include <audio/orchestra/api/Pulse.hpp>
 
-std::vector<std::string> audio::orchestra::Interface::getListApi() {
-	std::vector<std::string> apis;
+etk::Vector<etk::String> audio::orchestra::Interface::getListApi() {
+	etk::Vector<etk::String> apis;
 	// The order here will control the order of RtAudio's API search in
 	// the constructor.
 	for (size_t iii=0; iii<m_apiAvaillable.size(); ++iii) {
-		apis.push_back(m_apiAvaillable[iii].first);
+		apis.pushBack(m_apiAvaillable[iii].first);
 	}
 	return apis;
 }
 
 
 
-void audio::orchestra::Interface::openApi(const std::string& _api) {
+void audio::orchestra::Interface::openApi(const etk::String& _api) {
 	m_api.reset();
 	for (size_t iii=0; iii<m_apiAvaillable.size(); ++iii) {
 		ATA_INFO("try open " << m_apiAvaillable[iii].first);
 		if (_api == m_apiAvaillable[iii].first) {
 			ATA_INFO("    ==> call it");
 			m_api = m_apiAvaillable[iii].second();
-			if (m_api != nullptr) {
+			if (m_api != null) {
 				return;
 			}
 		}
@@ -49,7 +48,7 @@ void audio::orchestra::Interface::openApi(const std::string& _api) {
 
 
 audio::orchestra::Interface::Interface() :
-  m_api(nullptr) {
+  m_api(null) {
 	ATA_DEBUG("Add interface:");
 #if defined(ORCHESTRA_BUILD_JACK)
 	addInterface(audio::orchestra::typeJack, audio::orchestra::api::Jack::create);
@@ -80,13 +79,13 @@ audio::orchestra::Interface::Interface() :
 #endif
 }
 
-void audio::orchestra::Interface::addInterface(const std::string& _api, ememory::SharedPtr<Api> (*_callbackCreate)()) {
-	m_apiAvaillable.push_back(std::pair<std::string, ememory::SharedPtr<Api> (*)()>(_api, _callbackCreate));
+void audio::orchestra::Interface::addInterface(const etk::String& _api, ememory::SharedPtr<Api> (*_callbackCreate)()) {
+	m_apiAvaillable.pushBack(etk::Pair<etk::String, ememory::SharedPtr<Api> (*)()>(_api, _callbackCreate));
 }
 
 enum audio::orchestra::error audio::orchestra::Interface::clear() {
 	ATA_INFO("Clear API ...");
-	if (m_api == nullptr) {
+	if (m_api == null) {
 		ATA_WARNING("Interface NOT started!");
 		return audio::orchestra::error_none;
 	}
@@ -94,9 +93,9 @@ enum audio::orchestra::error audio::orchestra::Interface::clear() {
 	return audio::orchestra::error_none;
 }
 
-enum audio::orchestra::error audio::orchestra::Interface::instanciate(const std::string& _api) {
+enum audio::orchestra::error audio::orchestra::Interface::instanciate(const etk::String& _api) {
 	ATA_INFO("Instanciate API ...");
-	if (m_api != nullptr) {
+	if (m_api != null) {
 		ATA_WARNING("Interface already started!");
 		return audio::orchestra::error_none;
 	}
@@ -104,7 +103,7 @@ enum audio::orchestra::error audio::orchestra::Interface::instanciate(const std:
 		ATA_INFO("API specified : " << _api);
 		// Attempt to open the specified API.
 		openApi(_api);
-		if (m_api != nullptr) {
+		if (m_api != null) {
 			if (m_api->getDeviceCount() != 0) {
 				ATA_INFO("    ==> api open");
 			}
@@ -118,12 +117,12 @@ enum audio::orchestra::error audio::orchestra::Interface::instanciate(const std:
 	ATA_INFO("Auto choice API :");
 	// Iterate through the compiled APIs and return as soon as we find
 	// one with at least one device or we reach the end of the list.
-	std::vector<std::string> apis = getListApi();
+	etk::Vector<etk::String> apis = getListApi();
 	ATA_INFO(" find : " << apis.size() << " apis.");
 	for (size_t iii=0; iii<apis.size(); ++iii) {
 		ATA_INFO("try open ...");
 		openApi(apis[iii]);
-		if(m_api == nullptr) {
+		if(m_api == null) {
 			ATA_ERROR("    ==> can not create ...");
 			continue;
 		}
@@ -134,7 +133,7 @@ enum audio::orchestra::error audio::orchestra::Interface::instanciate(const std:
 			ATA_INFO("    ==> Interface exist, but have no devices: " << m_api->getDeviceCount());
 		}
 	}
-	if (m_api != nullptr) {
+	if (m_api != null) {
 		return audio::orchestra::error_none;
 	}
 	ATA_ERROR("API NOT Supported '" << _api << "' not in " << getListApi());
@@ -153,7 +152,7 @@ enum audio::orchestra::error audio::orchestra::Interface::openStream(audio::orch
                                                                      uint32_t* _bufferFrames,
                                                                      audio::orchestra::AirTAudioCallback _callback,
                                                                      const audio::orchestra::StreamOptions& _options) {
-	if (m_api == nullptr) {
+	if (m_api == null) {
 		return audio::orchestra::error_inputNull;
 	}
 	return m_api->openStream(_outputParameters,
@@ -166,12 +165,12 @@ enum audio::orchestra::error audio::orchestra::Interface::openStream(audio::orch
 }
 
 bool audio::orchestra::Interface::isMasterOf(audio::orchestra::Interface& _interface) {
-	if (m_api == nullptr) {
-		ATA_ERROR("Current Master API is nullptr ...");
+	if (m_api == null) {
+		ATA_ERROR("Current Master API is null ...");
 		return false;
 	}
-	if (_interface.m_api == nullptr) {
-		ATA_ERROR("Current Slave API is nullptr ...");
+	if (_interface.m_api == null) {
+		ATA_ERROR("Current Slave API is null ...");
 		return false;
 	}
 	if (m_api->getCurrentApi() != _interface.m_api->getCurrentApi()) {

audio/orchestra/Interface.hpp

@@ -6,11 +6,12 @@
  */
 #pragma once
 
-#include <string>
-#include <vector>
-#include <audio/orchestra/base.h>
-#include <audio/orchestra/CallbackInfo.h>
-#include <audio/orchestra/Api.h>
+#include <etk/String.hpp>
+#include <etk/Vector.hpp>
+#include <etk/Pair.hpp>
+#include <audio/orchestra/base.hpp>
+#include <audio/orchestra/CallbackInfo.hpp>
+#include <audio/orchestra/Api.hpp>
 
 namespace audio {
 	namespace orchestra {
@@ -26,12 +27,12 @@ namespace audio {
 		 */
 		class Interface {
 			protected:
-				std::vector<std::pair<std::string, ememory::SharedPtr<Api> (*)()> > m_apiAvaillable;
+				etk::Vector<etk::Pair<etk::String, ememory::SharedPtr<Api> (*)()> > m_apiAvaillable;
 			protected:
 				ememory::SharedPtr<audio::orchestra::Api> m_api;
 			public:
-				void setName(const std::string& _name) {
-					if (m_api == nullptr) {
+				void setName(const etk::String& _name) {
+					if (m_api == null) {
 						return;
 					}
 					m_api->setName(_name);
@@ -40,13 +41,13 @@ namespace audio {
 				 * @brief Get the list of all availlable API in the system.
 				 * @return the list of all APIs
 				 */
-				std::vector<std::string> getListApi();
+				etk::Vector<etk::String> getListApi();
 				/**
 				 * @brief Add an interface of the Possible List.
 				 * @param[in] _api Type of the interface.
 				 * @param[in] _callbackCreate API creation callback.
 				 */
-				void addInterface(const std::string& _api, ememory::SharedPtr<Api> (*_callbackCreate)());
+				void addInterface(const etk::String& _api, ememory::SharedPtr<Api> (*_callbackCreate)());
 				/**
 				 * @brief The class constructor.
 				 * @note the creating of the basic instance is done by Instanciate
@@ -66,12 +67,12 @@ namespace audio {
 				/**
 				 * @brief Create an interface instance
 				 */
-				enum audio::orchestra::error instanciate(const std::string& _api = audio::orchestra::typeUndefined);
+				enum audio::orchestra::error instanciate(const etk::String& _api = audio::orchestra::typeUndefined);
 				/**
 				 * @return the audio API specifier for the current instance of airtaudio.
 				 */
-				const std::string& getCurrentApi() {
-					if (m_api == nullptr) {
+				const etk::String& getCurrentApi() {
+					if (m_api == null) {
 						return audio::orchestra::typeUndefined;
 					}
 					return m_api->getCurrentApi();
@@ -84,7 +85,7 @@ namespace audio {
 				 * a system error occurs during processing, a warning will be issued. 
 				 */
 				uint32_t getDeviceCount() {
-					if (m_api == nullptr) {
+					if (m_api == null) {
 						return 0;
 					}
 					return m_api->getDeviceCount();
@@ -101,13 +102,13 @@ namespace audio {
 				 * @return An audio::orchestra::DeviceInfo structure for a specified device number.
 				 */
 				audio::orchestra::DeviceInfo getDeviceInfo(uint32_t _device) {
-					if (m_api == nullptr) {
+					if (m_api == null) {
 						return audio::orchestra::DeviceInfo();
 					}
 					return m_api->getDeviceInfo(_device);
 				}
-				audio::orchestra::DeviceInfo getDeviceInfo(const std::string& _deviceName) {
-					if (m_api == nullptr) {
+				audio::orchestra::DeviceInfo getDeviceInfo(const etk::String& _deviceName) {
+					if (m_api == null) {
 						return audio::orchestra::DeviceInfo();
 					}
 					audio::orchestra::DeviceInfo info;
@@ -124,7 +125,7 @@ namespace audio {
 				 * before attempting to open a stream.
 				 */
 				uint32_t getDefaultOutputDevice() {
-					if (m_api == nullptr) {
+					if (m_api == null) {
 						return 0;
 					}
 					return m_api->getDefaultOutputDevice();
@@ -139,7 +140,7 @@ namespace audio {
 				 * before attempting to open a stream.
 				 */
 				uint32_t getDefaultInputDevice() {
-					if (m_api == nullptr) {
+					if (m_api == null) {
 						return 0;
 					}
 					return m_api->getDefaultInputDevice();
@@ -154,12 +155,12 @@ namespace audio {
 				 * @param _outputParameters Specifies output stream parameters to use
 				 *           when opening a stream, including a device ID, number of channels,
 				 *           and starting channel number. For input-only streams, this
-				 *           argument should be nullptr. The device ID is an index value between
+				 *           argument should be null. The device ID is an index value between
 				 *           0 and getDeviceCount() - 1.
 				 * @param  _inputParameters Specifies input stream parameters to use
 				 *           when opening a stream, including a device ID, number of channels,
 				 *           and starting channel number. For output-only streams, this
-				 *           argument should be nullptr. The device ID is an index value between
+				 *           argument should be null. The device ID is an index value between
 				 *           0 and getDeviceCount() - 1.
 				 * @param  _format An audio::format specifying the desired sample data format.
 				 * @param  _sampleRate The desired sample rate (sample frames per second).
@@ -183,12 +184,12 @@ namespace audio {
 				 *           when an error has occured.
 				 */
 				enum audio::orchestra::error openStream(audio::orchestra::StreamParameters *_outputParameters,
-				                                 audio::orchestra::StreamParameters *_inputParameters,
-				                                 enum audio::format _format,
-				                                 uint32_t _sampleRate,
-				                                 uint32_t* _bufferFrames,
-				                                 audio::orchestra::AirTAudioCallback _callback,
-				                                 const audio::orchestra::StreamOptions& _options = audio::orchestra::StreamOptions());
+				                                        audio::orchestra::StreamParameters *_inputParameters,
+				                                        enum audio::format _format,
+				                                        uint32_t _sampleRate,
+				                                        uint32_t* _bufferFrames,
+				                                        audio::orchestra::AirTAudioCallback _callback,
+				                                        const audio::orchestra::StreamOptions& _options = audio::orchestra::StreamOptions());
 				
 				/**
 				 * @brief A function that closes a stream and frees any associated stream memory.
@@ -197,7 +198,7 @@ namespace audio {
 				 * returns (no exception is thrown).
 				 */
 				enum audio::orchestra::error closeStream() {
-					if (m_api == nullptr) {
+					if (m_api == null) {
 						return audio::orchestra::error_inputNull;
 					}
 					return m_api->closeStream();
@@ -211,7 +212,7 @@ namespace audio {
 				 * running.
 				 */
 				enum audio::orchestra::error startStream() {
-					if (m_api == nullptr) {
+					if (m_api == null) {
 						return audio::orchestra::error_inputNull;
 					}
 					return m_api->startStream();
@@ -225,7 +226,7 @@ namespace audio {
 				 * stopped.
 				*/
 				enum audio::orchestra::error stopStream() {
-					if (m_api == nullptr) {
+					if (m_api == null) {
 						return audio::orchestra::error_inputNull;
 					}
 					return m_api->stopStream();
@@ -238,7 +239,7 @@ namespace audio {
 				 * stopped.
 				 */
 				enum audio::orchestra::error abortStream() {
-					if (m_api == nullptr) {
+					if (m_api == null) {
 						return audio::orchestra::error_inputNull;
 					}
 					return m_api->abortStream();
@@ -247,7 +248,7 @@ namespace audio {
 				 * @return true if a stream is open and false if not.
 				 */
 				bool isStreamOpen() const {
-					if (m_api == nullptr) {
+					if (m_api == null) {
 						return false;
 					}
 					return m_api->isStreamOpen();
@@ -256,7 +257,7 @@ namespace audio {
 				 * @return true if the stream is running and false if it is stopped or not open.
 				 */
 				bool isStreamRunning() const {
-					if (m_api == nullptr) {
+					if (m_api == null) {
 						return false;
 					}
 					return m_api->isStreamRunning();
@@ -266,7 +267,7 @@ namespace audio {
 				 * @return the number of elapsed seconds since the stream was started.
 				 */
 				audio::Time getStreamTime() {
-					if (m_api == nullptr) {
+					if (m_api == null) {
 						return audio::Time();
 					}
 					return m_api->getStreamTime();
@@ -281,7 +282,7 @@ namespace audio {
 				 * @return The internal stream latency in sample frames.
 				 */
 				long getStreamLatency() {
-					if (m_api == nullptr) {
+					if (m_api == null) {
 						return 0;
 					}
 					return m_api->getStreamLatency();
@@ -293,14 +294,14 @@ namespace audio {
 				 * @return Returns actual sample rate in use by the stream.
 				 */
 				uint32_t getStreamSampleRate() {
-					if (m_api == nullptr) {
+					if (m_api == null) {
 						return 0;
 					}
 					return m_api->getStreamSampleRate();
 				}
 				bool isMasterOf(audio::orchestra::Interface& _interface);
 			protected:
-				void openApi(const std::string& _api);
+				void openApi(const etk::String& _api);
 		};
 	}
 }

audio/orchestra/StreamOptions.cpp

@@ -5,9 +5,9 @@
  * @fork from RTAudio
  */
 
-#include <audio/orchestra/StreamOptions.h>
-#include <etk/stdTools.h>
-#include <audio/orchestra/debug.h>
+#include <audio/orchestra/StreamOptions.hpp>
+#include <etk/stdTools.hpp>
+#include <audio/orchestra/debug.hpp>
 
 static const char* listValue[] = {
 	"hardware",
@@ -15,13 +15,13 @@ static const char* listValue[] = {
 	"soft"
 };
 
-std::ostream& audio::orchestra::operator <<(std::ostream& _os, enum audio::orchestra::timestampMode _obj) {
+etk::Stream& audio::orchestra::operator <<(etk::Stream& _os, enum audio::orchestra::timestampMode _obj) {
 	_os << listValue[_obj];
 	return _os;
 }
 
 namespace etk {
-	template <> bool from_string<enum audio::orchestra::timestampMode>(enum audio::orchestra::timestampMode& _variableRet, const std::string& _value) {
+	template <> bool from_string<enum audio::orchestra::timestampMode>(enum audio::orchestra::timestampMode& _variableRet, const etk::String& _value) {
 		if (_value == "hardware") {
 			_variableRet = audio::orchestra::timestampMode_Hardware;
 			return true;
@@ -37,7 +37,7 @@ namespace etk {
 		return false;
 	}
 	
-	template <enum audio::orchestra::timestampMode> std::string to_string(const enum audio::orchestra::timestampMode& _variable) {
+	template <enum audio::orchestra::timestampMode> etk::String toString(const enum audio::orchestra::timestampMode& _variable) {
 		return listValue[_variable];
 	}
 }

audio/orchestra/StreamOptions.hpp

@@ -6,7 +6,8 @@
  */
 #pragma once
 
-#include <audio/orchestra/Flags.h>
+#include <audio/orchestra/Flags.hpp>
+#include <etk/String.hpp>
 
 namespace audio {
 	namespace orchestra {
@@ -15,13 +16,13 @@ namespace audio {
 			timestampMode_trigered, //!< get harware triger time stamp and increment with duration
 			timestampMode_soft, //!< Simulate all timestamp.
 		};
-		std::ostream& operator <<(std::ostream& _os, enum audio::orchestra::timestampMode _obj);
+		etk::Stream& operator <<(etk::Stream& _os, enum audio::orchestra::timestampMode _obj);
 		
 		class StreamOptions {
 			public:
 				audio::orchestra::Flags flags; //!< A bit-mask of stream flags
 				uint32_t numberOfBuffers; //!< Number of stream buffers.
-				std::string streamName; //!< A stream name (currently used only in Jack).
+				etk::String streamName; //!< A stream name (currently used only in Jack).
 				enum timestampMode mode; //!< mode of timestamping data...
 				// Default constructor.
 				StreamOptions() :

audio/orchestra/StreamParameters.hpp

@@ -14,7 +14,7 @@ namespace audio {
 		class StreamParameters {
 			public:
 				int32_t deviceId; //!< Device index (-1 to getDeviceCount() - 1).
-				std::string deviceName; //!< name of the device (if deviceId==-1 this must not be == "", and the oposite ...)
+				etk::String deviceName; //!< name of the device (if deviceId==-1 this must not be == "", and the oposite ...)
 				uint32_t nChannels; //!< Number of channels.
 				uint32_t firstChannel; //!< First channel index on device (default = 0).
 				// Default constructor.

audio/orchestra/api/Alsa.cpp

@@ -9,27 +9,27 @@
 #if defined(ORCHESTRA_BUILD_ALSA)
 
 #include <alsa/asoundlib.h>
-#include <unistd.h>
-#include <audio/orchestra/Interface.h>
-#include <audio/orchestra/debug.h>
-#include <etk/stdTools.h>
-#include <ethread/tools.h>
-#include <limits.h>
-#include <audio/orchestra/api/Alsa.h>
+
+#include <audio/orchestra/Interface.hpp>
+#include <audio/orchestra/debug.hpp>
+#include <etk/stdTools.hpp>
+#include <ethread/tools.hpp>
+#include <audio/orchestra/api/Alsa.hpp>
 extern "C" {
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <sched.h>
-#include <errno.h>
-#include <getopt.h>
-#include <sys/time.h>
-#include <math.h>
-#include <poll.h>
+	#include <sched.h>
+	#include <getopt.h>
+	#include <sys/time.h>
+	#include <poll.h>
+	#include <errno.h>
+	#include <stdio.h>
+	#include <stdlib.h>
+	#include <string.h>
+	#include <math.h>
+	#include <limits.h>
 }
 
 ememory::SharedPtr<audio::orchestra::Api> audio::orchestra::api::Alsa::create() {
-	return ememory::SharedPtr<audio::orchestra::api::Alsa>(new audio::orchestra::api::Alsa());
+	return ememory::SharedPtr<audio::orchestra::api::Alsa>(ETK_NEW(audio::orchestra::api::Alsa));
 }
 
 namespace audio {
@@ -39,17 +39,17 @@ namespace audio {
 				public:
 					snd_pcm_t *handle;
 					bool xrun[2];
-					std::condition_variable runnable_cv;
+					ethread::Semaphore m_semaphore;
 					bool runnable;
-					std::thread* thread;
+					ethread::Thread* thread;
 					bool threadRunning;
 					bool mmapInterface; //!< enable or disable mmap mode...
 					enum timestampMode timeMode; //!< the timestamp of the flow came from the harware.
-					std::vector<snd_pcm_channel_area_t> areas;
+					etk::Vector<snd_pcm_channel_area_t> areas;
 					AlsaPrivate() :
-					  handle(nullptr),
+					  handle(null),
 					  runnable(false),
-					  thread(nullptr),
+					  thread(null),
 					  threadRunning(false),
 					  mmapInterface(false),
 					  timeMode(timestampMode_soft) {
@@ -112,7 +112,7 @@ nextcard:
 	return nDevices;
 }
 
-bool audio::orchestra::api::Alsa::getNamedDeviceInfoLocal(const std::string& _deviceName, audio::orchestra::DeviceInfo& _info, int32_t _cardId, int32_t _subdevice, int32_t _localDeviceId, bool _input) {
+bool audio::orchestra::api::Alsa::getNamedDeviceInfoLocal(const etk::String& _deviceName, audio::orchestra::DeviceInfo& _info, int32_t _cardId, int32_t _subdevice, int32_t _localDeviceId, bool _input) {
 	int32_t result;
 	snd_ctl_t *chandle;
 	int32_t openMode = SND_PCM_ASYNC;
@@ -130,7 +130,7 @@ bool audio::orchestra::api::Alsa::getNamedDeviceInfoLocal(const std::string& _de
 		stream = SND_PCM_STREAM_PLAYBACK;
 	}
 	snd_pcm_info_set_stream(pcminfo, stream);
-	std::vector<std::string> listElement = etk::split(_deviceName, ',');
+	etk::Vector<etk::String> listElement = etk::split(_deviceName, ',');
 	if (listElement.size() == 0) {
 		ATA_ERROR("can not get control interface = '" << _deviceName << "' Can not plit at ',' ...");
 		return false;
@@ -190,15 +190,13 @@ bool audio::orchestra::api::Alsa::getNamedDeviceInfoLocal(const std::string& _de
 	}
 	ATA_DEBUG("Input channel = " << value);
 	for (int32_t iii=0; iii<value; ++iii) {
-		_info.channels.push_back(audio::channel_unknow);
+		_info.channels.pushBack(audio::channel_unknow);
 	}
 	// Test our discrete set of sample rate values.
 	_info.sampleRates.clear();
-	for (std::vector<uint32_t>::const_iterator it(audio::orchestra::genericSampleRate().begin()); 
-	     it != audio::orchestra::genericSampleRate().end();
-	     ++it ) {
-		if (snd_pcm_hw_params_test_rate(phandle, params, *it, 0) == 0) {
-			_info.sampleRates.push_back(*it);
+	for (auto &it: audio::orchestra::genericSampleRate()) {
+		if (snd_pcm_hw_params_test_rate(phandle, params, it, 0) == 0) {
+			_info.sampleRates.pushBack(it);
 		}
 	}
 	if (_info.sampleRates.size() == 0) {
@@ -211,27 +209,27 @@ bool audio::orchestra::api::Alsa::getNamedDeviceInfoLocal(const std::string& _de
 	_info.nativeFormats.clear();
 	format = SND_PCM_FORMAT_S8;
 	if (snd_pcm_hw_params_test_format(phandle, params, format) == 0) {
-		_info.nativeFormats.push_back(audio::format_int8);
+		_info.nativeFormats.pushBack(audio::format_int8);
 	}
 	format = SND_PCM_FORMAT_S16;
 	if (snd_pcm_hw_params_test_format(phandle, params, format) == 0) {
-		_info.nativeFormats.push_back(audio::format_int16);
+		_info.nativeFormats.pushBack(audio::format_int16);
 	}
 	format = SND_PCM_FORMAT_S24;
 	if (snd_pcm_hw_params_test_format(phandle, params, format) == 0) {
-		_info.nativeFormats.push_back(audio::format_int24);
+		_info.nativeFormats.pushBack(audio::format_int24);
 	}
 	format = SND_PCM_FORMAT_S32;
 	if (snd_pcm_hw_params_test_format(phandle, params, format) == 0) {
-		_info.nativeFormats.push_back(audio::format_int32);
+		_info.nativeFormats.pushBack(audio::format_int32);
 	}
 	format = SND_PCM_FORMAT_FLOAT;
 	if (snd_pcm_hw_params_test_format(phandle, params, format) == 0) {
-		_info.nativeFormats.push_back(audio::format_float);
+		_info.nativeFormats.pushBack(audio::format_float);
 	}
 	format = SND_PCM_FORMAT_FLOAT64;
 	if (snd_pcm_hw_params_test_format(phandle, params, format) == 0) {
-		_info.nativeFormats.push_back(audio::format_double);
+		_info.nativeFormats.pushBack(audio::format_double);
 	}
 	// Check that we have at least one supported format
 	if (_info.nativeFormats.size() == 0) {
@@ -406,7 +404,7 @@ foundDevice:
 	return openName(name, _mode, _channels, _firstChannel, _sampleRate, _format, _bufferSize, _options);
 }
 
-bool audio::orchestra::api::Alsa::openName(const std::string& _deviceName,
+bool audio::orchestra::api::Alsa::openName(const etk::String& _deviceName,
                                            audio::orchestra::mode _mode,
                                            uint32_t _channels,
                                            uint32_t _firstChannel,
@@ -676,7 +674,7 @@ bool audio::orchestra::api::Alsa::openName(const std::string& _deviceName,
 	}
 	snd_pcm_uframes_t val;
 	// Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
-	snd_pcm_sw_params_t *swParams = nullptr;
+	snd_pcm_sw_params_t *swParams = null;
 	snd_pcm_sw_params_alloca(&swParams);
 	snd_pcm_sw_params_current(m_private->handle, swParams);
 	#if 0
@@ -783,10 +781,10 @@ bool audio::orchestra::api::Alsa::openName(const std::string& _deviceName,
 		bufferBytes *= *_bufferSize;
 		if (m_deviceBuffer) {
 			free(m_deviceBuffer);
-			m_deviceBuffer = nullptr;
+			m_deviceBuffer = null;
 		}
 		m_deviceBuffer = (char *) calloc(bufferBytes, 1);
-		if (m_deviceBuffer == nullptr) {
+		if (m_deviceBuffer == null) {
 			ATA_ERROR("error allocating device buffer memory.");
 			goto error;
 		}
@@ -803,8 +801,8 @@ bool audio::orchestra::api::Alsa::openName(const std::string& _deviceName,
 	// Setup callback thread.
 	m_private->threadRunning = true;
 	ATA_INFO("create thread ...");
-	m_private->thread = new std::thread(&audio::orchestra::api::Alsa::alsaCallbackEvent, this);
-	if (m_private->thread == nullptr) {
+	m_private->thread = ETK_NEW(ethread::Thread, [=]() {callbackEvent();});
+	if (m_private->thread == null) {
 		m_private->threadRunning = false;
 		ATA_ERROR("creating callback thread!");
 		goto error;
@@ -814,7 +812,7 @@ bool audio::orchestra::api::Alsa::openName(const std::string& _deviceName,
 error:
 	if (m_private->handle) {
 		snd_pcm_close(m_private->handle);
-		m_private->handle = nullptr;
+		m_private->handle = null;
 	}
 	for (int32_t iii=0; iii<2; ++iii) {
 		m_userBuffer[iii].clear();
@@ -836,12 +834,12 @@ enum audio::orchestra::error audio::orchestra::api::Alsa::closeStream() {
 	m_mutex.lock();
 	if (m_state == audio::orchestra::state::stopped) {
 		m_private->runnable = true;
-		m_private->runnable_cv.notify_one();
+		m_private->m_semaphore.post();
 	}
-	m_mutex.unlock();
-	if (m_private->thread != nullptr) {
+	m_mutex.unLock();
+	if (m_private->thread != null) {
 		m_private->thread->join();
-		m_private->thread = nullptr;
+		m_private->thread = null;
 	}
 	if (m_state == audio::orchestra::state::running) {
 		m_state = audio::orchestra::state::stopped;
@@ -850,7 +848,7 @@ enum audio::orchestra::error audio::orchestra::api::Alsa::closeStream() {
 	// close all stream :
 	if (m_private->handle) {
 		snd_pcm_close(m_private->handle);
-		m_private->handle = nullptr;
+		m_private->handle = null;
 	}
 	for (int32_t iii=0; iii<2; ++iii) {
 		m_userBuffer[iii].clear();
@@ -865,23 +863,29 @@ enum audio::orchestra::error audio::orchestra::api::Alsa::closeStream() {
 }
 
 enum audio::orchestra::error audio::orchestra::api::Alsa::startStream() {
+	ATA_DEBUG("Start stream (DEGIN)");
 	// TODO : Check return ...
 	//audio::orchestra::Api::startStream();
 	// This method calls snd_pcm_prepare if the device isn't already in that state.
 	if (verifyStream() != audio::orchestra::error_none) {
+		ATA_WARNING("the stream not prepared!");
 		return audio::orchestra::error_fail;
 	}
 	if (m_state == audio::orchestra::state::running) {
 		ATA_ERROR("the stream is already running!");
 		return audio::orchestra::error_warning;
 	}
-	std::unique_lock<std::mutex> lck(m_mutex);
+	ATA_DEBUG("Lock");
+	ethread::UniqueLock lck(m_mutex);
+	ATA_DEBUG("Lock (done)");
 	int32_t result = 0;
 	snd_pcm_state_t state;
-	if (m_private->handle == nullptr) {
-		ATA_ERROR("send nullptr to alsa ...");
+	if (m_private->handle == null) {
+		ATA_ERROR("send null to alsa ...");
 	}
+	ATA_DEBUG("snd_pcm_state");
 	state = snd_pcm_state(m_private->handle);
+	ATA_DEBUG("snd_pcm_state (done)");
 	if (state != SND_PCM_STATE_PREPARED) {
 		ATA_ERROR("prepare stream");
 		result = snd_pcm_prepare(m_private->handle);
@@ -893,10 +897,12 @@ enum audio::orchestra::error audio::orchestra::api::Alsa::startStream() {
 	m_state = audio::orchestra::state::running;
 unlock:
 	m_private->runnable = true;
-	m_private->runnable_cv.notify_one();
+	m_private->m_semaphore.post();
 	if (result >= 0) {
+		ATA_DEBUG("Start stream (END2)");
 		return audio::orchestra::error_none;
 	}
+	ATA_DEBUG("Start stream (END)");
 	return audio::orchestra::error_systemError;
 }
 
@@ -909,7 +915,7 @@ enum audio::orchestra::error audio::orchestra::api::Alsa::stopStream() {
 		return audio::orchestra::error_warning;
 	}
 	m_state = audio::orchestra::state::stopped;
-	std::unique_lock<std::mutex> lck(m_mutex);
+	ethread::UniqueLock lck(m_mutex);
 	int32_t result = 0;
 	if (m_mode == audio::orchestra::mode_output) {
 		result = snd_pcm_drain(m_private->handle);
@@ -936,7 +942,7 @@ enum audio::orchestra::error audio::orchestra::api::Alsa::abortStream() {
 		return audio::orchestra::error_warning;
 	}
 	m_state = audio::orchestra::state::stopped;
-	std::unique_lock<std::mutex> lck(m_mutex);
+	ethread::UniqueLock lck(m_mutex);
 	int32_t result = 0;
 	result = snd_pcm_drop(m_private->handle);
 	if (result < 0) {
@@ -950,12 +956,6 @@ unlock:
 	return audio::orchestra::error_systemError;
 }
 
-
-void audio::orchestra::api::Alsa::alsaCallbackEvent(void *_userData) {
-	audio::orchestra::api::Alsa* myClass = reinterpret_cast<audio::orchestra::api::Alsa*>(_userData);
-	myClass->callbackEvent();
-}
-
 /**
  * @briefTransfer method - write and wait for room in buffer using poll
  */
@@ -979,9 +979,8 @@ static int32_t wait_for_poll(snd_pcm_t* _handle, struct pollfd* _ufds, unsigned
 void audio::orchestra::api::Alsa::callbackEvent() {
 	// Lock while the system is not started ...
 	if (m_state == audio::orchestra::state::stopped) {
-		std::unique_lock<std::mutex> lck(m_mutex);
 		while (!m_private->runnable) {
-			m_private->runnable_cv.wait(lck);
+			m_private->m_semaphore.wait();
 		}
 		if (m_state != audio::orchestra::state::running) {
 			return;
@@ -989,7 +988,7 @@ void audio::orchestra::api::Alsa::callbackEvent() {
 	}
 	ethread::setName("Alsa IO-" + m_name);
 	//Wait data with poll
-	std::vector<struct pollfd> ufds;
+	etk::Vector<struct pollfd> ufds;
 	signed short *ptr;
 	int32_t err, count, cptr, init;
 	count = snd_pcm_poll_descriptors_count(m_private->handle);
@@ -1032,7 +1031,7 @@ void audio::orchestra::api::Alsa::callbackEvent() {
 audio::Time audio::orchestra::api::Alsa::getStreamTime() {
 	//ATA_DEBUG("mode : " << m_private->timeMode);
 	if (m_private->timeMode == timestampMode_Hardware) {
-		snd_pcm_status_t *status = nullptr;
+		snd_pcm_status_t *status = null;
 		snd_pcm_status_alloca(&status);
 		// get harware timestamp all the time:
 		snd_pcm_status(m_private->handle, status);
@@ -1055,7 +1054,7 @@ audio::Time audio::orchestra::api::Alsa::getStreamTime() {
 		ATA_VERBOSE("snd_pcm_status_get_htstamp : " << m_startTime);
 		snd_pcm_sframes_t delay = snd_pcm_status_get_delay(status);
 		audio::Duration timeDelay = audio::Duration(0, delay*1000000000LL/int64_t(m_sampleRate));
-		ATA_VERBOSE("delay : " << timeDelay.count() << " ns");
+		ATA_VERBOSE("delay : " << timeDelay);
 		//return m_startTime + m_duration;
 		if (m_mode == audio::orchestra::mode_output) {
 			// output
@@ -1067,7 +1066,7 @@ audio::Time audio::orchestra::api::Alsa::getStreamTime() {
 		return m_startTime;
 	} else if (m_private->timeMode == timestampMode_trigered) {
 		if (m_startTime == audio::Time()) {
-			snd_pcm_status_t *status = nullptr;
+			snd_pcm_status_t *status = null;
 			snd_pcm_status_alloca(&status);
 			// get harware timestamp all the time:
 			snd_pcm_status(m_private->handle, status);
@@ -1106,9 +1105,9 @@ void audio::orchestra::api::Alsa::callbackEventOneCycleRead() {
 	}
 	int32_t doStopStream = 0;
 	audio::Time streamTime;
-	std::vector<enum audio::orchestra::status> status;
+	etk::Vector<enum audio::orchestra::status> status;
 	if (m_private->xrun[0] == true) {
-		status.push_back(audio::orchestra::status::underflow);
+		status.pushBack(audio::orchestra::status::underflow);
 		m_private->xrun[0] = false;
 	}
 	int32_t result;
@@ -1121,7 +1120,7 @@ void audio::orchestra::api::Alsa::callbackEventOneCycleRead() {
 		// !!! goto unlock;
 	}
 	
-	std::unique_lock<std::mutex> lck(m_mutex);
+	ethread::UniqueLock lck(m_mutex);
 	// Setup parameters.
 	if (m_doConvertBuffer[1]) {
 		buffer = m_deviceBuffer;
@@ -1167,7 +1166,7 @@ void audio::orchestra::api::Alsa::callbackEventOneCycleRead() {
 			}
 		} else {
 			ATA_ERROR("audio read error, " << snd_strerror(result) << ".");
-			usleep(10000);
+			ethread::sleepMilliSeconds((10));
 		}
 		// TODO : Notify application ... audio::orchestra::error_warning;
 		goto noInput;
@@ -1193,7 +1192,7 @@ noInput:
 		audio::Time startCall = audio::Time::now();
 		doStopStream = m_callback(&m_userBuffer[1][0],
 		                          streamTime,// - audio::Duration(m_latency[1]*1000000000LL/int64_t(m_sampleRate)),
-		                          nullptr,
+		                          null,
 		                          audio::Time(),
 		                          m_bufferSize,
 		                          status);
@@ -1201,7 +1200,7 @@ noInput:
 		audio::Duration timeDelay(0, m_bufferSize*1000000000LL/int64_t(m_sampleRate));
 		audio::Duration timeProcess = stopCall - startCall;
 		if (timeDelay <= timeProcess) {
-			ATA_ERROR("SOFT XRUN ... : (bufferTime) " << timeDelay.count() << " < " << timeProcess.count() << " (process time) ns");
+			ATA_ERROR("SOFT XRUN ... : (bufferTime) " << timeDelay << " < " << timeProcess << " (process time)");
 		}
 	}
 	if (doStopStream == 2) {
@@ -1223,9 +1222,9 @@ void audio::orchestra::api::Alsa::callbackEventOneCycleWrite() {
 	}
 	int32_t doStopStream = 0;
 	audio::Time streamTime;
-	std::vector<enum audio::orchestra::status> status;
+	etk::Vector<enum audio::orchestra::status> status;
 	if (m_private->xrun[1] == true) {
-		status.push_back(audio::orchestra::status::overflow);
+		status.pushBack(audio::orchestra::status::overflow);
 		m_private->xrun[1] = false;
 	}
 	int32_t result;
@@ -1241,7 +1240,7 @@ void audio::orchestra::api::Alsa::callbackEventOneCycleWrite() {
 	streamTime = getStreamTime();
 	{
 		audio::Time startCall = audio::Time::now();
-		doStopStream = m_callback(nullptr,
+		doStopStream = m_callback(null,
 		                          audio::Time(),
 		                          &m_userBuffer[0][0],
 		                          streamTime,// + audio::Duration(m_latency[0]*1000000000LL/int64_t(m_sampleRate)),
@@ -1251,14 +1250,14 @@ void audio::orchestra::api::Alsa::callbackEventOneCycleWrite() {
 		audio::Duration timeDelay(0, m_bufferSize*1000000000LL/int64_t(m_sampleRate));
 		audio::Duration timeProcess = stopCall - startCall;
 		if (timeDelay <= timeProcess) {
-			ATA_ERROR("SOFT XRUN ... : (bufferTime) " << timeDelay.count() << " < " << timeProcess.count() << " (process time) ns");
+			ATA_ERROR("SOFT XRUN ... : (bufferTime) " << timeDelay << " < " << timeProcess << " (process time)");
 		}
 	}
 	if (doStopStream == 2) {
 		abortStream();
 		return;
 	}
-	std::unique_lock<std::mutex> lck(m_mutex);
+	ethread::UniqueLock lck(m_mutex);
 	// Setup parameters and do buffer conversion if necessary.
 	if (m_doConvertBuffer[0]) {
 		buffer = m_deviceBuffer;
@@ -1325,9 +1324,9 @@ void audio::orchestra::api::Alsa::callbackEventOneCycleMMAPWrite() {
 	}
 	int32_t doStopStream = 0;
 	audio::Time streamTime;
-	std::vector<enum audio::orchestra::status> status;
+	etk::Vector<enum audio::orchestra::status> status;
 	if (m_private->xrun[1] == true) {
-		status.push_back(audio::orchestra::status::overflow);
+		status.pushBack(audio::orchestra::status::overflow);
 		m_private->xrun[1] = false;
 	}
 	int32_t result;
@@ -1347,7 +1346,7 @@ void audio::orchestra::api::Alsa::callbackEventOneCycleMMAPWrite() {
 	streamTime = getStreamTime();
 	{
 		audio::Time startCall = audio::Time::now();
-		doStopStream = m_callback(nullptr,
+		doStopStream = m_callback(null,
 		                          audio::Time(),
 		                          &m_userBuffer[0][0],
 		                          streamTime,// + audio::Duration(m_latency[0]*1000000000LL/int64_t(m_sampleRate)),
@@ -1357,7 +1356,7 @@ void audio::orchestra::api::Alsa::callbackEventOneCycleMMAPWrite() {
 		audio::Duration timeDelay(0, m_bufferSize*1000000000LL/int64_t(m_sampleRate));
 		audio::Duration timeProcess = stopCall - startCall;
 		if (timeDelay <= timeProcess) {
-			ATA_ERROR("SOFT XRUN ... : (bufferTime) " << timeDelay.count() << " < " << timeProcess.count() << " (process time) ns");
+			ATA_ERROR("SOFT XRUN ... : (bufferTime) " << timeDelay << " < " << timeProcess << " (process time)");
 		}
 	}
 	if (doStopStream == 2) {
@@ -1365,7 +1364,7 @@ void audio::orchestra::api::Alsa::callbackEventOneCycleMMAPWrite() {
 		return;
 	}
 	{
-		std::unique_lock<std::mutex> lck(m_mutex);
+		ethread::UniqueLock lck(m_mutex);
 		// Setup parameters and do buffer conversion if necessary.
 		if (m_doConvertBuffer[0]) {
 			buffer = m_deviceBuffer;
@@ -1397,7 +1396,7 @@ void audio::orchestra::api::Alsa::callbackEventOneCycleMMAPWrite() {
 			// TODO: Understand why this does not work ...
 			// Write samples to device in interleaved/non-interleaved format.
 			if (m_deviceInterleaved[0]) {
-				const snd_pcm_channel_area_t* myAreas = nullptr;
+				const snd_pcm_channel_area_t* myAreas = null;
 				snd_pcm_uframes_t offset, frames;
 				frames = m_bufferSize;
 				ATA_DEBUG("START");
@@ -1459,9 +1458,9 @@ void audio::orchestra::api::Alsa::callbackEventOneCycleMMAPRead() {
 	}
 	int32_t doStopStream = 0;
 	audio::Time streamTime;
-	std::vector<enum audio::orchestra::status> status;
+	etk::Vector<enum audio::orchestra::status> status;
 	if (m_private->xrun[0] == true) {
-		status.push_back(audio::orchestra::status::underflow);
+		status.pushBack(audio::orchestra::status::underflow);
 		m_private->xrun[0] = false;
 	}
 	int32_t result;
@@ -1474,7 +1473,7 @@ void audio::orchestra::api::Alsa::callbackEventOneCycleMMAPRead() {
 		goto unlock;
 	}
 	{
-		std::unique_lock<std::mutex> lck(m_mutex);
+		ethread::UniqueLock lck(m_mutex);
 		// Setup parameters.
 		if (m_doConvertBuffer[1]) {
 			buffer = m_deviceBuffer;
@@ -1521,7 +1520,7 @@ void audio::orchestra::api::Alsa::callbackEventOneCycleMMAPRead() {
 				}
 			} else {
 				ATA_ERROR("audio read error, " << snd_strerror(result) << ".");
-				usleep(10000);
+				ethread::sleepMilliSeconds((10));
 			}
 			// TODO : Notify application ... audio::orchestra::error_warning;
 			goto noInput;
@@ -1549,7 +1548,7 @@ noInput:
 		audio::Time startCall = audio::Time::now();
 		doStopStream = m_callback(&m_userBuffer[1][0],
 		                          streamTime,// - audio::Duration(m_latency[1]*1000000000LL/int64_t(m_sampleRate)),
-		                          nullptr,
+		                          null,
 		                          audio::Time(),
 		                          m_bufferSize,
 		                          status);
@@ -1557,7 +1556,7 @@ noInput:
 		audio::Duration timeDelay(0, m_bufferSize*1000000000LL/int64_t(m_sampleRate));
 		audio::Duration timeProcess = stopCall - startCall;
 		if (timeDelay <= timeProcess) {
-			ATA_ERROR("SOFT XRUN ... : (bufferTime) " << timeDelay.count() << " < " << timeProcess.count() << " (process time) ns");
+			ATA_ERROR("SOFT XRUN ... : (bufferTime) " << timeDelay << " < " << timeProcess << " (process time) ns");
 		}
 	}
 	if (doStopStream == 2) {
@@ -1574,7 +1573,7 @@ unlock:
 
 bool audio::orchestra::api::Alsa::isMasterOf(ememory::SharedPtr<audio::orchestra::Api> _api) {
 	ememory::SharedPtr<audio::orchestra::api::Alsa> slave = ememory::dynamicPointerCast<audio::orchestra::api::Alsa>(_api);
-	if (slave == nullptr) {
+	if (slave == null) {
 		ATA_ERROR("NULL ptr API (not ALSA ...)");
 		return false;
 	}
@@ -1586,11 +1585,11 @@ bool audio::orchestra::api::Alsa::isMasterOf(ememory::SharedPtr<audio::orchestra
 		ATA_ERROR("The SLAVE stream is already running! ==> can not synchronize ...");
 		return false;
 	}
-	snd_pcm_t * master = nullptr;
-	if (m_private->handle != nullptr) {
+	snd_pcm_t * master = null;
+	if (m_private->handle != null) {
 		master = m_private->handle;
 	}
-	if (master == nullptr) {
+	if (master == null) {
 		ATA_ERROR("No ALSA handles ...");
 		return false;
 	}

audio/orchestra/api/Alsa.hpp

@@ -18,19 +18,19 @@ namespace audio {
 				public:
 					Alsa();
 					virtual ~Alsa();
-					const std::string& getCurrentApi() {
+					const etk::String& getCurrentApi() {
 						return audio::orchestra::typeAlsa;
 					}
 					uint32_t getDeviceCount();
 				private:
-					bool getNamedDeviceInfoLocal(const std::string& _deviceName,
+					bool getNamedDeviceInfoLocal(const etk::String& _deviceName,
 					                             audio::orchestra::DeviceInfo& _info,
 					                             int32_t _cardId=-1, // Alsa card ID
 					                             int32_t _subdevice=-1, // alsa subdevice ID
 					                             int32_t _localDeviceId=-1,// local ID of device find
 					                             bool _input=false);
 				public:
-					bool getNamedDeviceInfo(const std::string& _deviceName, audio::orchestra::DeviceInfo& _info) {
+					bool getNamedDeviceInfo(const etk::String& _deviceName, audio::orchestra::DeviceInfo& _info) {
 						return getNamedDeviceInfoLocal(_deviceName, _info);
 					}
 					audio::orchestra::DeviceInfo getDeviceInfo(uint32_t _device);
@@ -47,11 +47,9 @@ namespace audio {
 					void callbackEventOneCycleWrite();
 					void callbackEventOneCycleMMAPRead();
 					void callbackEventOneCycleMMAPWrite();
-				private:
-					static void alsaCallbackEvent(void* _userData);
 				private:
 					ememory::SharedPtr<AlsaPrivate> m_private;
-					std::vector<audio::orchestra::DeviceInfo> m_devices;
+					etk::Vector<audio::orchestra::DeviceInfo> m_devices;
 					void saveDeviceInfo();
 					bool open(uint32_t _device,
 					          enum audio::orchestra::mode _mode,
@@ -62,7 +60,7 @@ namespace audio {
 					          uint32_t *_bufferSize,
 					          const audio::orchestra::StreamOptions& _options);
 					
-					bool openName(const std::string& _deviceName,
+					bool openName(const etk::String& _deviceName,
 					              audio::orchestra::mode _mode,
 					              uint32_t _channels,
 					              uint32_t _firstChannel,

audio/orchestra/api/Android.cpp

@@ -8,16 +8,18 @@
 #ifdef ORCHESTRA_BUILD_JAVA
 
 //#include <ewol/context/Context.h>
-#include <unistd.h>
-#include <audio/orchestra/Interface.h>
-#include <audio/orchestra/debug.h>
-#include <audio/orchestra/api/AndroidNativeInterface.h>
-#include <audio/orchestra/api/Android.h>
-#include <limits.h>
+
+#include <audio/orchestra/Interface.hpp>
+#include <audio/orchestra/debug.hpp>
+#include <audio/orchestra/api/AndroidNativeInterface.hpp>
+#include <audio/orchestra/api/Android.hpp>
+extern "C" {
+	#include <limits.h>
+}
 
 ememory::SharedPtr<audio::orchestra::Api> audio::orchestra::api::Android::create() {
 	ATA_INFO("Create Android device ... ");
-	return ememory::SharedPtr<audio::orchestra::api::Android>(new audio::orchestra::api::Android());
+	return ememory::SharedPtr<audio::orchestra::api::Android>(ETK_NEW(audio::orchestra::api::Android));
 }
 
 
@@ -69,15 +71,15 @@ enum audio::orchestra::error audio::orchestra::api::Android::abortStream() {
 
 void audio::orchestra::api::Android::playback(int16_t* _dst, int32_t _nbChunk) {
 	// clear output buffer:
-	if (_dst != nullptr) {
+	if (_dst != null) {
 		memset(_dst, 0, _nbChunk*audio::getFormatBytes(m_deviceFormat[modeToIdTable(m_mode)])*m_nDeviceChannels[modeToIdTable(m_mode)]);
 	}
 	int32_t doStopStream = 0;
 	audio::Time streamTime = getStreamTime();
-	std::vector<enum audio::orchestra::status> status;
+	etk::Vector<enum audio::orchestra::status> status;
 	if (m_doConvertBuffer[modeToIdTable(m_mode)] == true) {
 		ATA_VERBOSE("Need playback data " << int32_t(_nbChunk) << " userbuffer size = " << m_userBuffer[audio::orchestra::mode_output].size() << "pointer=" << int64_t(&m_userBuffer[audio::orchestra::mode_output][0]));
-		doStopStream = m_callback(nullptr,
+		doStopStream = m_callback(null,
 		                          audio::Time(),
 		                          &m_userBuffer[m_mode][0],
 		                          streamTime,
@@ -86,7 +88,7 @@ void audio::orchestra::api::Android::playback(int16_t* _dst, int32_t _nbChunk) {
 		convertBuffer((char*)_dst, (char*)&m_userBuffer[audio::orchestra::mode_output][0], m_convertInfo[audio::orchestra::mode_output]);
 	} else {
 		ATA_VERBOSE("Need playback data " << int32_t(_nbChunk) << " pointer=" << int64_t(_dst));
-		doStopStream = m_callback(nullptr,
+		doStopStream = m_callback(null,
 		                          audio::Time(),
 		                          _dst,
 		                          streamTime,
@@ -104,13 +106,13 @@ void audio::orchestra::api::Android::playback(int16_t* _dst, int32_t _nbChunk) {
 void audio::orchestra::api::Android::record(int16_t* _dst, int32_t _nbChunk) {
 	int32_t doStopStream = 0;
 	audio::Time streamTime = getStreamTime();
-	std::vector<enum audio::orchestra::status> status;
+	etk::Vector<enum audio::orchestra::status> status;
 	if (m_doConvertBuffer[modeToIdTable(m_mode)] == true) {
 		ATA_VERBOSE("Need playback data " << int32_t(_nbChunk) << " userbuffer size = " << m_userBuffer[audio::orchestra::mode_output].size() << "pointer=" << int64_t(&m_userBuffer[audio::orchestra::mode_output][0]));
 		convertBuffer((char*)&m_userBuffer[audio::orchestra::mode_input][0], (char*)_dst, m_convertInfo[audio::orchestra::mode_input]);
 		doStopStream = m_callback(&m_userBuffer[m_mode][0],
 		                          streamTime,
-		                          nullptr,
+		                          null,
 		                          audio::Time(),
 		                          uint32_t(_nbChunk),
 		                          status);
@@ -118,7 +120,7 @@ void audio::orchestra::api::Android::record(int16_t* _dst, int32_t _nbChunk) {
 		ATA_VERBOSE("Need playback data " << int32_t(_nbChunk) << " pointer=" << int64_t(_dst));
 		doStopStream = m_callback(_dst,
 		                          streamTime,
-		                          nullptr,
+		                          null,
 		                          audio::Time(),
 		                          uint32_t(_nbChunk),
 		                          status);

audio/orchestra/api/Android.hpp

@@ -7,7 +7,7 @@
 
 #ifdef ORCHESTRA_BUILD_JAVA
 
-#include <audio/orchestra/Interface.h>
+#include <audio/orchestra/Interface.hpp>
 
 namespace audio {
 	namespace orchestra {
@@ -18,7 +18,7 @@ namespace audio {
 				public:
 					Android();
 					virtual ~Android();
-					const std::string& getCurrentApi() {
+					const etk::String& getCurrentApi() {
 						return audio::orchestra::typeJava;
 					}
 					uint32_t getDeviceCount();
@@ -39,7 +39,7 @@ namespace audio {
 						return m_uid;
 					}
 				private:
-					std::vector<audio::orchestra::DeviceInfo> m_devices;
+					etk::Vector<audio::orchestra::DeviceInfo> m_devices;
 					void saveDeviceInfo();
 					bool open(uint32_t _device,
 					          audio::orchestra::mode _mode,

audio/orchestra/api/AndroidNativeInterface.cpp

@@ -4,18 +4,20 @@
  * @license APACHE v2.0 (see license file)
  */
 
-#include <jni.h>
-#include <pthread.h>
-#include <mutex>
-#include <audio/orchestra/debug.h>
-#include <audio/orchestra/error.h>
-#include <audio/orchestra/api/AndroidNativeInterface.h>
-#include <audio/orchestra/api/Android.h>
+extern "C" {
+	#include <jni.h>
+	#include <pthread.h>
+}
+#include <ethread/Mutex.hpp>
+#include <audio/orchestra/debug.hpp>
+#include <audio/orchestra/error.hpp>
+#include <audio/orchestra/api/AndroidNativeInterface.hpp>
+#include <audio/orchestra/api/Android.hpp>
 /* include auto generated file */
 #include <org_musicdsp_orchestra_OrchestraConstants.h>
-#include <jvm-basics/jvm-basics.h>
-#include <ememory/memory.h>
-#include <ejson/ejson.h>
+#include <jvm-basics/jvm-basics.hpp>
+#include <ememory/memory.hpp>
+#include <ejson/ejson.hpp>
 
 class AndroidOrchestraContext {
 	public:
@@ -35,19 +37,19 @@ class AndroidOrchestraContext {
 	private:
 		bool safeInitMethodID(jmethodID& _mid, jclass& _cls, const char* _name, const char* _sign) {
 			_mid = m_JavaVirtualMachinePointer->GetMethodID(_cls, _name, _sign);
-			if(_mid == nullptr) {
+			if(_mid == null) {
 				ATA_ERROR("C->java : Can't find the method " << _name);
 				/* remove access on the virtual machine : */
-				m_JavaVirtualMachinePointer = nullptr;
+				m_JavaVirtualMachinePointer = null;
 				return false;
 			}
 			return true;
 		}
 		bool java_attach_current_thread(int *_rstatus) {
 			ATA_DEBUG("C->java : call java");
-			if (jvm_basics::getJavaVM() == nullptr) {
+			if (jvm_basics::getJavaVM() == null) {
 				ATA_ERROR("C->java : JVM not initialised");
-				m_JavaVirtualMachinePointer = nullptr;
+				m_JavaVirtualMachinePointer = null;
 				return false;
 			}
 			*_rstatus = jvm_basics::getJavaVM()->GetEnv((void **) &m_JavaVirtualMachinePointer, JNI_VERSION_1_6);
@@ -55,12 +57,12 @@ class AndroidOrchestraContext {
 				JavaVMAttachArgs lJavaVMAttachArgs;
 				lJavaVMAttachArgs.version = JNI_VERSION_1_6;
 				lJavaVMAttachArgs.name = "EwolNativeThread";
-				lJavaVMAttachArgs.group = nullptr; 
+				lJavaVMAttachArgs.group = null; 
 				int status = jvm_basics::getJavaVM()->AttachCurrentThread(&m_JavaVirtualMachinePointer, &lJavaVMAttachArgs);
 				jvm_basics::checkExceptionJavaVM(m_JavaVirtualMachinePointer);
 				if (status != JNI_OK) {
 					ATA_ERROR("C->java : AttachCurrentThread failed : " << status);
-					m_JavaVirtualMachinePointer = nullptr;
+					m_JavaVirtualMachinePointer = null;
 					return false;
 				}
 			}
@@ -69,13 +71,13 @@ class AndroidOrchestraContext {
 		void java_detach_current_thread(int _status) {
 			if(_status == JNI_EDETACHED) {
 				jvm_basics::getJavaVM()->DetachCurrentThread();
-				m_JavaVirtualMachinePointer = nullptr;
+				m_JavaVirtualMachinePointer = null;
 			}
 		}
 		
 	public:
 		AndroidOrchestraContext(JNIEnv* _env, jclass _classBase, jobject _objCallback) :
-		  m_JavaVirtualMachinePointer(nullptr),
+		  m_JavaVirtualMachinePointer(null),
 		  m_javaClassOrchestra(0),
 		  m_javaClassOrchestraCallback(0),
 		  m_javaObjectOrchestraCallback(0),
@@ -93,7 +95,7 @@ class AndroidOrchestraContext {
 			ATA_DEBUG("*******************************************");
 			m_JavaVirtualMachinePointer = _env;
 			// get default needed all time elements : 
-			if (m_JavaVirtualMachinePointer == nullptr) {
+			if (m_JavaVirtualMachinePointer == null) {
 				ATA_ERROR("C->java: NULLPTR jvm interface");
 				return;
 			}
@@ -102,15 +104,15 @@ class AndroidOrchestraContext {
 			if (m_javaClassOrchestra == 0) {
 				ATA_ERROR("C->java : Can't find org/musicdsp/orchestra/OrchestraNative class");
 				// remove access on the virtual machine : 
-				m_JavaVirtualMachinePointer = nullptr;
+				m_JavaVirtualMachinePointer = null;
 				return;
 			}
 			/* The object field extends Activity and implement OrchestraCallback */
 			m_javaClassOrchestraCallback = m_JavaVirtualMachinePointer->GetObjectClass(_objCallback);
-			if(m_javaClassOrchestraCallback == nullptr) {
+			if(m_javaClassOrchestraCallback == null) {
 				ATA_ERROR("C->java : Can't find org/musicdsp/orchestra/OrchestraManagerCallback class");
 				// remove access on the virtual machine : 
-				m_JavaVirtualMachinePointer = nullptr;
+				m_JavaVirtualMachinePointer = null;
 				return;
 			}
 			bool functionCallbackIsMissing = false;
@@ -183,7 +185,7 @@ class AndroidOrchestraContext {
 			
 			
 			m_javaObjectOrchestraCallback = _env->NewGlobalRef(_objCallback);
-			if (m_javaObjectOrchestraCallback == nullptr) {
+			if (m_javaObjectOrchestraCallback == null) {
 				functionCallbackIsMissing = true;
 			}
 			
@@ -191,7 +193,7 @@ class AndroidOrchestraContext {
 			if (m_javaDefaultClassString == 0) {
 				ATA_ERROR("C->java : Can't find java/lang/String" );
 				// remove access on the virtual machine : 
-				m_JavaVirtualMachinePointer = nullptr;
+				m_JavaVirtualMachinePointer = null;
 				functionCallbackIsMissing = true;
 			}
 			if (functionCallbackIsMissing == true) {
@@ -205,7 +207,7 @@ class AndroidOrchestraContext {
 		
 		void unInit(JNIEnv* _env) {
 			_env->DeleteGlobalRef(m_javaObjectOrchestraCallback);
-			m_javaObjectOrchestraCallback = nullptr;
+			m_javaObjectOrchestraCallback = null;
 		}
 		
 		uint32_t getDeviceCount() {
@@ -234,8 +236,8 @@ class AndroidOrchestraContext {
 			}
 			//Call java ...
 			jstring returnString = (jstring) m_JavaVirtualMachinePointer->CallObjectMethod(m_javaObjectOrchestraCallback, m_javaMethodOrchestraActivityAudioGetDeviceProperty, _idDevice);
-			const char *js = m_JavaVirtualMachinePointer->GetStringUTFChars(returnString, nullptr);
-			std::string retString(js);
+			const char *js = m_JavaVirtualMachinePointer->GetStringUTFChars(returnString, null);
+			etk::String retString(js);
 			m_JavaVirtualMachinePointer->ReleaseStringUTFChars(returnString, js);
 			//m_JavaVirtualMachinePointer->DeleteLocalRef(returnString);
 			// manage execption : 
@@ -247,36 +249,36 @@ class AndroidOrchestraContext {
 				
 				return info;
 			}
-			info.name = doc.getStringValue("name", "no-name");
-			if (doc.getStringValue("type", "output") == "output") {
+			info.name = doc["name"].toString().get("no-name");
+			if (doc["type"].toString().get("output") == "output") {
 				info.input = false;
 			} else {
 				info.input = true;
 			}
-			ememory::SharedPtr<const ejson::Array> list = doc.getArray("sample-rate");
-			if (list != nullptr) {
-				for (size_t iii=0; iii<list->size(); ++iii) {
-					info.sampleRates.push_back(int32_t(list->getNumberValue(iii, 48000)));
+			ejson::Array list = doc["sample-rate"].toArray();
+			if (list.exist() == true) {
+				for (auto it : list) {
+					info.sampleRates.pushBack(int32_t(it.toNumber().get(48000)));
 				}
 			}
-			list = doc.getArray("channels");
-			if (list != nullptr) {
-				for (size_t iii=0; iii<list->size(); ++iii) {
-					info.channels.push_back(audio::getChannelFromString(list->getStringValue(iii, "???")));
+			list = doc["channels"].toArray();
+			if (list.exist() == true) {
+				for (auto it : list) {
+					info.channels.pushBack(audio::getChannelFromString(it.toString().get("???")));
 				}
 			}
-			list = doc.getArray("format");
-			if (list != nullptr) {
-				for (size_t iii=0; iii<list->size(); ++iii) {
-					info.nativeFormats.push_back(audio::getFormatFromString(list->getStringValue(iii, "???")));
+			list = doc["format"].toArray();
+			if (list.exist() == true) {
+				for (auto it : list) {
+					info.nativeFormats.pushBack(audio::getFormatFromString(it.toString().get("???")));
 				}
 			}
-			info.isDefault = doc.getBooleanValue("default", false);
+			info.isDefault = doc["default"].toBoolean().get(false);
 			info.isCorrect = true;
 			return info;
 		}
 	private:
-		std::vector<ememory::WeakPtr<audio::orchestra::api::Android> > m_instanceList; // list of connected handle ...
+		etk::Vector<ememory::WeakPtr<audio::orchestra::api::Android> > m_instanceList; // list of connected handle ...
 		//AndroidAudioCallback m_audioCallBack;
 		//void* m_audioCallBackUserData;
 	public:
@@ -305,7 +307,7 @@ class AndroidOrchestraContext {
 			jvm_basics::checkExceptionJavaVM(m_JavaVirtualMachinePointer);
 			java_detach_current_thread(status);
 			if (int32_t(ret) >= 0) {
-				m_instanceList.push_back(_instance);
+				m_instanceList.pushBack(_instance);
 				return int32_t(ret);
 			}
 			return -1;
@@ -369,7 +371,7 @@ class AndroidOrchestraContext {
 			auto it = m_instanceList.begin();
 			while (it != m_instanceList.end()) {
 				auto elem = it->lock();
-				if (elem == nullptr) {
+				if (elem == null) {
 					it = m_instanceList.erase(it);
 					continue;
 				}
@@ -383,7 +385,7 @@ class AndroidOrchestraContext {
 			auto it = m_instanceList.begin();
 			while (it != m_instanceList.end()) {
 				auto elem = it->lock();
-				if (elem == nullptr) {
+				if (elem == null) {
 					it = m_instanceList.erase(it);
 					continue;
 				}
@@ -400,7 +402,7 @@ static int32_t s_nbContextRequested(0);
 
 
 uint32_t audio::orchestra::api::android::getDeviceCount() {
-	if (s_localContext == nullptr) {
+	if (s_localContext == null) {
 		ATA_ERROR("Have no Orchertra API instanciate in JAVA ...");
 		return 0;
 	}
@@ -408,7 +410,7 @@ uint32_t audio::orchestra::api::android::getDeviceCount() {
 }
 
 audio::orchestra::DeviceInfo audio::orchestra::api::android::getDeviceInfo(uint32_t _device) {
-	if (s_localContext == nullptr) {
+	if (s_localContext == null) {
 		return audio::orchestra::DeviceInfo();
 	}
 	return s_localContext->getDeviceInfo(_device);
@@ -423,35 +425,35 @@ int32_t audio::orchestra::api::android::open(uint32_t _device,
                                              uint32_t *_bufferSize,
                                              const audio::orchestra::StreamOptions& _options,
                                              ememory::SharedPtr<audio::orchestra::api::Android> _instance) {
-	if (s_localContext == nullptr) {
+	if (s_localContext == null) {
 		return -1;
 	}
 	return s_localContext->open(_device, _mode, _channels, _firstChannel, _sampleRate, _format, _bufferSize, _options, _instance);
 }
 
 enum audio::orchestra::error audio::orchestra::api::android::closeStream(int32_t _id) {
-	if (s_localContext == nullptr) {
+	if (s_localContext == null) {
 		return audio::orchestra::error_fail;
 	}
 	return s_localContext->closeStream(_id);
 }
 
 enum audio::orchestra::error audio::orchestra::api::android::startStream(int32_t _id) {
-	if (s_localContext == nullptr) {
+	if (s_localContext == null) {
 		return audio::orchestra::error_fail;
 	}
 	return s_localContext->startStream(_id);
 }
 
 enum audio::orchestra::error audio::orchestra::api::android::stopStream(int32_t _id) {
-	if (s_localContext == nullptr) {
+	if (s_localContext == null) {
 		return audio::orchestra::error_fail;
 	}
 	return s_localContext->stopStream(_id);
 }
 
 enum audio::orchestra::error audio::orchestra::api::android::abortStream(int32_t _id) {
-	if (s_localContext == nullptr) {
+	if (s_localContext == null) {
 		return audio::orchestra::error_fail;
 	}
 	return s_localContext->abortStream(_id);
@@ -461,15 +463,15 @@ extern "C" {
 	void Java_org_musicdsp_orchestra_OrchestraNative_NNsetJavaManager(JNIEnv* _env,
 	                                                                  jclass _classBase,
 	                                                                  jobject _objCallback) {
-		std::unique_lock<std::mutex> lock(jvm_basics::getMutexJavaVM());
+		ethread::UniqueLock lock(jvm_basics::getMutexJavaVM());
 		ATA_INFO("*******************************************");
 		ATA_INFO("** Creating Orchestra context            **");
 		ATA_INFO("*******************************************");
-		if (s_localContext != nullptr) {
+		if (s_localContext != null) {
 			s_nbContextRequested++;
 		}
 		s_localContext = ememory::makeShared<AndroidOrchestraContext>(_env, _classBase, _objCallback);
-		if (s_localContext == nullptr) {
+		if (s_localContext == null) {
 			ATA_ERROR("Can not allocate the orchestra main context instance");
 			return;
 		}
@@ -477,7 +479,7 @@ extern "C" {
 	}
 	
 	void Java_org_musicdsp_orchestra_OrchestraNative_NNsetJavaManagerRemove(JNIEnv* _env, jclass _cls) {
-		std::unique_lock<std::mutex> lock(jvm_basics::getMutexJavaVM());
+		ethread::UniqueLock lock(jvm_basics::getMutexJavaVM());
 		ATA_INFO("*******************************************");
 		ATA_INFO("** remove Orchestra Pointer              **");
 		ATA_INFO("*******************************************");
@@ -495,15 +497,15 @@ extern "C" {
 	                                                            jint _id,
 	                                                            jshortArray _location,
 	                                                            jint _nbChunk) {
-		std::unique_lock<std::mutex> lock(jvm_basics::getMutexJavaVM());
-		if (s_localContext == nullptr) {
+		ethread::UniqueLock lock(jvm_basics::getMutexJavaVM());
+		if (s_localContext == null) {
 			ATA_ERROR("Call audio with no more Low level interface");
 			return;
 		}
 		// get the short* pointer from the Java array
 		jboolean isCopy;
 		jshort* dst = _env->GetShortArrayElements(_location, &isCopy);
-		if (dst != nullptr) {
+		if (dst != null) {
 			//ATA_INFO("Need audioData " << int32_t(_nbChunk));
 			s_localContext->playback(int32_t(_id), static_cast<short*>(dst), int32_t(_nbChunk));
 		}
@@ -518,15 +520,15 @@ extern "C" {
 	                                                          jint _id,
 	                                                          jshortArray _location,
 	                                                          jint _nbChunk) {
-		std::unique_lock<std::mutex> lock(jvm_basics::getMutexJavaVM());
-		if (s_localContext == nullptr) {
+		ethread::UniqueLock lock(jvm_basics::getMutexJavaVM());
+		if (s_localContext == null) {
 			ATA_ERROR("Call audio with no more Low level interface");
 			return;
 		}
 		// get the short* pointer from the Java array
 		jboolean isCopy;
 		jshort* dst = _env->GetShortArrayElements(_location, &isCopy);
-		if (dst != nullptr) {
+		if (dst != null) {
 			//ATA_INFO("Need audioData " << int32_t(_nbChunk));
 			s_localContext->record(int32_t(_id), static_cast<short*>(dst), int32_t(_nbChunk));
 		}

audio/orchestra/api/AndroidNativeInterface.hpp

@@ -7,12 +7,12 @@
 
 #ifdef ORCHESTRA_BUILD_JAVA
 
-#include <audio/orchestra/DeviceInfo.h>
-#include <audio/orchestra/mode.h>
-#include <audio/orchestra/error.h>
-#include <audio/orchestra/StreamOptions.h>
-#include <audio/format.h>
-#include <ememory/memory.h>
+#include <audio/orchestra/DeviceInfo.hpp>
+#include <audio/orchestra/mode.hpp>
+#include <audio/orchestra/error.hpp>
+#include <audio/orchestra/StreamOptions.hpp>
+#include <audio/format.hpp>
+#include <ememory/memory.hpp>
 
 namespace audio {
 	namespace orchestra {

audio/orchestra/api/Asio.cpp

@@ -8,11 +8,11 @@
 
 #if defined(ORCHESTRA_BUILD_ASIO)
 
-#include <audio/orchestra/Interface.h>
-#include <audio/orchestra/debug.h>
+#include <audio/orchestra/Interface.hpp>
+#include <audio/orchestra/debug.hpp>
 
 ememory::SharedPtr<audio::orchestra::Api> audio::orchestra::api::Asio::create() {
-	return ememory::SharedPtr<audio::orchestra::api::Asio>(new audio::orchestra::api::Asio());
+	return ememory::SharedPtr<audio::orchestra::api::Asio>(ETK_NEW(audio::orchestra::api::Asio));
 }
 
 
@@ -32,12 +32,13 @@ ememory::SharedPtr<audio::orchestra::Api> audio::orchestra::api::Asio::create()
 // on information found in
 // http://www.cs.wustl.edu/~schmidt/win32-cv-1.html.
 
-#include "asiosys.h"
-#include "asio.h"
-#include "iasiothiscallresolver.h"
-#include "asiodrivers.h"
-#include <cmath>
-
+extern "C" {
+	#include "asiosys.h"
+	#include "asio.h"
+	#include "iasiothiscallresolver.h"
+	#include "asiodrivers.h"
+	#include <math.h>
+}
 static AsioDrivers drivers;
 static ASIOCallbacks asioCallbacks;
 static ASIODriverInfo driverInfo;
@@ -70,12 +71,12 @@ static void sampleRateChanged(ASIOSampleRate _sRate);
 static long asioMessages(long _selector, long _value, void* _message, double* _opt);
 
 audio::orchestra::api::Asio::Asio() :
-  m_private(new audio::orchestra::api::AsioPrivate()) {
+  m_private(ETK_NEW(audio::orchestra::api::AsioPrivate)) {
 	// ASIO cannot run on a multi-threaded appartment. You can call
 	// CoInitialize beforehand, but it must be for appartment threading
 	// (in which case, CoInitilialize will return S_FALSE here).
 	m_coInitialized = false;
-	HRESULT hr = CoInitialize(nullptr); 
+	HRESULT hr = CoInitialize(null); 
 	if (FAILED(hr)) {
 		ATA_ERROR("requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)");
 	}
@@ -154,7 +155,7 @@ rtaudio::DeviceInfo audio::orchestra::api::Asio::getDeviceInfo(uint32_t _device)
 	for (uint32_t i=0; i<MAX_SAMPLE_RATES; i++) {
 		result = ASIOCanSampleRate((ASIOSampleRate) SAMPLE_RATES[i]);
 		if (result == ASE_OK) {
-			info.sampleRates.push_back(SAMPLE_RATES[i]);
+			info.sampleRates.pushBack(SAMPLE_RATES[i]);
 		}
 	}
 	// Determine supported data types ... just check first channel and assume rest are the same.
@@ -173,19 +174,19 @@ rtaudio::DeviceInfo audio::orchestra::api::Asio::getDeviceInfo(uint32_t _device)
 	info.nativeFormats.clear();
 	if (    channelInfo.type == ASIOSTInt16MSB
 	     || channelInfo.type == ASIOSTInt16LSB) {
-		info.nativeFormats.push_back(audio::format_int16);
+		info.nativeFormats.pushBack(audio::format_int16);
 	} else if (    channelInfo.type == ASIOSTInt32MSB
 	            || channelInfo.type == ASIOSTInt32LSB) {
-		info.nativeFormats.push_back(audio::format_int32);
+		info.nativeFormats.pushBack(audio::format_int32);
 	} else if (    channelInfo.type == ASIOSTFloat32MSB
 	            || channelInfo.type == ASIOSTFloat32LSB) {
-		info.nativeFormats.push_back(audio::format_float);
+		info.nativeFormats.pushBack(audio::format_float);
 	} else if (    channelInfo.type == ASIOSTFloat64MSB
 	            || channelInfo.type == ASIOSTFloat64LSB) {
-		info.nativeFormats.push_back(audio::format_double);
+		info.nativeFormats.pushBack(audio::format_double);
 	} else if (    channelInfo.type == ASIOSTInt24MSB
 	            || channelInfo.type == ASIOSTInt24LSB) {
-		info.nativeFormats.push_back(audio::format_int24);
+		info.nativeFormats.pushBack(audio::format_int24);
 	}
 	if (info.outputChannels > 0){
 		if (getDefaultOutputDevice() == _device) {
@@ -378,10 +379,10 @@ bool audio::orchestra::api::Asio::open(uint32_t _device,
 				log2_of_max_size = i;
 			}
 		}
-		long min_delta = std::abs((long)*_bufferSize - ((long)1 << log2_of_min_size));
+		long min_delta = etk::abs((long)*_bufferSize - ((long)1 << log2_of_min_size));
 		int32_t min_delta_num = log2_of_min_size;
 		for (int32_t i = log2_of_min_size + 1; i <= log2_of_max_size; i++) {
-			long current_delta = std::abs((long)*_bufferSize - ((long)1 << i));
+			long current_delta = etk::abs((long)*_bufferSize - ((long)1 << i));
 			if (current_delta < min_delta) {
 				min_delta = current_delta;
 				min_delta_num = i;
@@ -410,10 +411,10 @@ bool audio::orchestra::api::Asio::open(uint32_t _device,
 	m_deviceInterleaved[modeToIdTable(_mode)] = false;
 	m_private->bufferInfos = 0;
 	// Create a manual-reset event.
-	m_private->condition = CreateEvent(nullptr,  // no security
+	m_private->condition = CreateEvent(null,  // no security
 	                                   TRUE,  // manual-reset
 	                                   FALSE, // non-signaled initially
-	                                   nullptr); // unnamed
+	                                   null); // unnamed
 	// Create the ASIO internal buffers.	Since RtAudio sets up input
 	// and output separately, we'll have to dispose of previously
 	// created output buffers for a duplex stream.
@@ -421,16 +422,16 @@ bool audio::orchestra::api::Asio::open(uint32_t _device,
 	if (    _mode == audio::orchestra::mode_input
 	     && m_mode == audio::orchestra::mode_output) {
 		ASIODisposeBuffers();
-		if (m_private->bufferInfos == nullptr) {
+		if (m_private->bufferInfos == null) {
 			free(m_private->bufferInfos);
-			m_private->bufferInfos = nullptr;
+			m_private->bufferInfos = null;
 		}
 	}
 	// Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
 	bool buffersAllocated = false;
 	uint32_t i, nChannels = m_nDeviceChannels[0] + m_nDeviceChannels[1];
 	m_private->bufferInfos = (ASIOBufferInfo *) malloc(nChannels * sizeof(ASIOBufferInfo));
-	if (m_private->bufferInfos == nullptr) {
+	if (m_private->bufferInfos == null) {
 		ATA_ERROR("error allocating bufferInfo memory for driver (" << driverName << ").");
 		goto error;
 	}
@@ -450,7 +451,7 @@ bool audio::orchestra::api::Asio::open(uint32_t _device,
 	asioCallbacks.bufferSwitch = &bufferSwitch;
 	asioCallbacks.sampleRateDidChange = &sampleRateChanged;
 	asioCallbacks.asioMessage = &asioMessages;
-	asioCallbacks.bufferSwitchTimeInfo = nullptr;
+	asioCallbacks.bufferSwitchTimeInfo = null;
 	result = ASIOCreateBuffers(m_private->bufferInfos, nChannels, m_bufferSize, &asioCallbacks);
 	if (result != ASE_OK) {
 		ATA_ERROR("driver (" << driverName << ") error (" << getAsioErrorString(result) << ") creating buffers.");
@@ -470,7 +471,7 @@ bool audio::orchestra::api::Asio::open(uint32_t _device,
 	uint64_t bufferBytes;
 	bufferBytes = m_nUserChannels[modeToIdTable(_mode)] * *_bufferSize * audio::getFormatBytes(m_userFormat);
 	m_userBuffer[modeToIdTable(_mode)] = (char *) calloc(bufferBytes, 1);
-	if (m_userBuffer[modeToIdTable(_mode)] == nullptr) {
+	if (m_userBuffer[modeToIdTable(_mode)] == null) {
 		ATA_ERROR("error allocating user buffer memory.");
 		goto error;
 	}
@@ -489,10 +490,10 @@ bool audio::orchestra::api::Asio::open(uint32_t _device,
 			bufferBytes *= *_bufferSize;
 			if (m_deviceBuffer) {
 				free(m_deviceBuffer);
-				m_deviceBuffer = nullptr;
+				m_deviceBuffer = null;
 			}
 			m_deviceBuffer = (char *) calloc(bufferBytes, 1);
-			if (m_deviceBuffer == nullptr) {
+			if (m_deviceBuffer == null) {
 				ATA_ERROR("error allocating device buffer memory.");
 				goto error;
 			}
@@ -529,9 +530,9 @@ error:
 	}
 	drivers.removeCurrentDriver();
 	CloseHandle(m_private->condition);
-	if (m_private->bufferInfos != nullptr) {
+	if (m_private->bufferInfos != null) {
 		free(m_private->bufferInfos);
-		m_private->bufferInfos = nullptr;
+		m_private->bufferInfos = null;
 	}
 	for (int32_t i=0; i<2; i++) {
 		if (m_userBuffer[i]) {
@@ -679,7 +680,7 @@ bool audio::orchestra::api::Asio::callbackEvent(long bufferIndex) {
 			SetEvent(m_private->condition);
 		} else { // spawn a thread to stop the stream
 			unsigned threadId;
-			m_callbackInfo.thread = _beginthreadex(nullptr,
+			m_callbackInfo.thread = _beginthreadex(null,
 			                                       0,
 			                                       &asioStopStream,
 			                                       &m_callbackInfo,
@@ -692,13 +693,13 @@ bool audio::orchestra::api::Asio::callbackEvent(long bufferIndex) {
 	// draining stream.
 	if (m_private->drainCounter == 0) {
 		audio::Time streamTime = getStreamTime();
-		std::vector<enum audio::orchestra::status status;
+		etk::Vector<enum audio::orchestra::status status;
 		if (m_mode != audio::orchestra::mode_input && asioXRun == true) {
-			status.push_back(audio::orchestra::status::underflow);
+			status.pushBack(audio::orchestra::status::underflow);
 			asioXRun = false;
 		}
 		if (m_mode != audio::orchestra::mode_output && asioXRun == true) {
-			status.push_back(audio::orchestra::status::underflow;
+			status.pushBack(audio::orchestra::status::underflow;
 			asioXRun = false;
 		}
 		int32_t cbReturnValue = info->callback(m_userBuffer[1],
@@ -711,7 +712,7 @@ bool audio::orchestra::api::Asio::callbackEvent(long bufferIndex) {
 			m_state = audio::orchestra::state::stopping;
 			m_private->drainCounter = 2;
 			unsigned threadId;
-			m_callbackInfo.thread = _beginthreadex(nullptr,
+			m_callbackInfo.thread = _beginthreadex(null,
 			                                              0,
 			                                              &asioStopStream,
 			                                              &m_callbackInfo,

audio/orchestra/api/Asio.hpp

@@ -17,7 +17,7 @@ namespace audio {
 				public:
 					Asio();
 					virtual ~Asio();
-					const std::string& getCurrentApi() {
+					const etk::String& getCurrentApi() {
 						return audio::orchestra::typeAsio;
 					}
 					uint32_t getDeviceCount();
@@ -34,7 +34,7 @@ namespace audio {
 					bool callbackEvent(long _bufferIndex);
 				private:
 					ememory::SharedPtr<AsioPrivate> m_private;
-					std::vector<audio::orchestra::DeviceInfo> m_devices;
+					etk::Vector<audio::orchestra::DeviceInfo> m_devices;
 					void saveDeviceInfo();
 					bool m_coInitialized;
 					bool open(uint32_t _device,

audio/orchestra/api/Core.cpp

@@ -14,14 +14,14 @@
 
 #if defined(__MACOSX_CORE__) || defined(ORCHESTRA_BUILD_MACOSX_CORE)
 
-#include <audio/orchestra/Interface.h>
-#include <audio/orchestra/debug.h>
-#include <thread>
-#include <ethread/tools.h>
-#include <audio/orchestra/api/Core.h>
+#include <audio/orchestra/Interface.hpp>
+#include <audio/orchestra/debug.hpp>
+#include <ethread/Thread.hpp>
+#include <ethread/tools.hpp>
+#include <audio/orchestra/api/Core.hpp>
 
 ememory::SharedPtr<audio::orchestra::Api> audio::orchestra::api::Core::create() {
-	return ememory::SharedPtr<audio::orchestra::api::Core>(new audio::orchestra::api::Core());
+	return ememory::SharedPtr<audio::orchestra::api::Core>(ETK_NEW(audio::orchestra::api::Core));
 }
 
 namespace audio {
@@ -37,7 +37,7 @@ namespace audio {
 					uint32_t nStreams[2]; // number of streams to use
 					bool xrun[2];
 					char *deviceBuffer;
-					std::condition_variable condition;
+					ethread::Semaphore m_semaphore;
 					int32_t drainCounter; // Tracks callback counts when draining
 					bool internalDrain; // Indicates if stop is initiated from callback or not.
 					CorePrivate() :
@@ -57,13 +57,13 @@ namespace audio {
 }
 
 audio::orchestra::api::Core::Core() :
-  m_private(new audio::orchestra::api::CorePrivate()) {
+  m_private(ETK_NEW(audio::orchestra::api::CorePrivate)) {
 #if defined(AVAILABLE_MAC_OS_X_VERSION_10_6_AND_LATER)
 	// This is a largely undocumented but absolutely necessary
 	// requirement starting with OS-X 10.6.	If not called, queries and
 	// updates to various audio device properties are not handled
 	// correctly.
-	CFRunLoopRef theRunLoop = nullptr;
+	CFRunLoopRef theRunLoop = null;
 	AudioObjectPropertyAddress property = {
 		kAudioHardwarePropertyRunLoop,
 		kAudioObjectPropertyScopeGlobal,
@@ -72,7 +72,7 @@ audio::orchestra::api::Core::Core() :
 	OSStatus result = AudioObjectSetPropertyData(kAudioObjectSystemObject,
 	                                             &property,
 	                                             0,
-	                                             nullptr,
+	                                             null,
 	                                             sizeof(CFRunLoopRef),
 	                                             &theRunLoop);
 	if (result != noErr) {
@@ -98,7 +98,7 @@ uint32_t audio::orchestra::api::Core::getDeviceCount() {
 		kAudioObjectPropertyScopeGlobal,
 		kAudioObjectPropertyElementMaster
 	};
-	OSStatus result = AudioObjectGetPropertyDataSize(kAudioObjectSystemObject, &propertyAddress, 0, nullptr, &dataSize);
+	OSStatus result = AudioObjectGetPropertyDataSize(kAudioObjectSystemObject, &propertyAddress, 0, null, &dataSize);
 	if (result != noErr) {
 		ATA_ERROR("OS-X error getting device info!");
 		return 0;
@@ -121,7 +121,7 @@ uint32_t audio::orchestra::api::Core::getDefaultInputDevice() {
 	OSStatus result = AudioObjectGetPropertyData(kAudioObjectSystemObject,
 	                                             &property,
 	                                             0,
-	                                             nullptr,
+	                                             null,
 	                                             &dataSize,
 	                                             &id);
 	if (result != noErr) {
@@ -134,7 +134,7 @@ uint32_t audio::orchestra::api::Core::getDefaultInputDevice() {
 	result = AudioObjectGetPropertyData(kAudioObjectSystemObject,
 	                                    &property,
 	                                    0,
-	                                    nullptr,
+	                                    null,
 	                                    &dataSize,
 	                                    (void*)&deviceList);
 	if (result != noErr) {
@@ -165,7 +165,7 @@ uint32_t audio::orchestra::api::Core::getDefaultOutputDevice() {
 	OSStatus result = AudioObjectGetPropertyData(kAudioObjectSystemObject,
 	                                             &property,
 	                                             0,
-	                                             nullptr,
+	                                             null,
 	                                             &dataSize,
 	                                             &id);
 	if (result != noErr) {
@@ -178,7 +178,7 @@ uint32_t audio::orchestra::api::Core::getDefaultOutputDevice() {
 	result = AudioObjectGetPropertyData(kAudioObjectSystemObject,
 	                                    &property,
 	                                    0,
-	                                    nullptr,
+	                                    null,
 	                                    &dataSize,
 	                                    (void*)&deviceList);
 	if (result != noErr) {
@@ -223,7 +223,7 @@ audio::orchestra::DeviceInfo audio::orchestra::api::Core::getDeviceInfo(uint32_t
 	OSStatus result = AudioObjectGetPropertyData(kAudioObjectSystemObject,
 	                                             &property,
 	                                             0,
-	                                             nullptr,
+	                                             null,
 	                                             &dataSize,
 	                                             (void*)&deviceList);
 	if (result != noErr) {
@@ -239,7 +239,7 @@ audio::orchestra::DeviceInfo audio::orchestra::api::Core::getDeviceInfo(uint32_t
 	CFStringRef cfname;
 	dataSize = sizeof(CFStringRef);
 	property.mSelector = kAudioObjectPropertyManufacturer;
-	result = AudioObjectGetPropertyData(id, &property, 0, nullptr, &dataSize, &cfname);
+	result = AudioObjectGetPropertyData(id, &property, 0, null, &dataSize, &cfname);
 	if (result != noErr) {
 		ATA_ERROR("system error (" << getErrorCode(result) << ") getting device manufacturer.");
 		info.clear();
@@ -247,14 +247,14 @@ audio::orchestra::DeviceInfo audio::orchestra::api::Core::getDeviceInfo(uint32_t
 	}
 	//const char *mname = CFStringGetCStringPtr(cfname, CFStringGetSystemEncoding());
 	int32_t length = CFStringGetLength(cfname);
-	std::vector<char> name;
+	etk::Vector<char> name;
 	name.resize(length * 3 + 1, '\0');
 	CFStringGetCString(cfname, &name[0], length * 3 + 1, CFStringGetSystemEncoding());
 	info.name.append(&name[0], strlen(&name[0]));
 	info.name.append(": ");
 	CFRelease(cfname);
 	property.mSelector = kAudioObjectPropertyName;
-	result = AudioObjectGetPropertyData(id, &property, 0, nullptr, &dataSize, &cfname);
+	result = AudioObjectGetPropertyData(id, &property, 0, null, &dataSize, &cfname);
 	if (result != noErr) {
 		ATA_ERROR("system error (" << getErrorCode(result) << ") getting device name.");
 		info.clear();
@@ -276,9 +276,9 @@ audio::orchestra::DeviceInfo audio::orchestra::api::Core::getDeviceInfo(uint32_t
 	} else {
 		property.mScope = kAudioDevicePropertyScopeInput;
 	}
-	AudioBufferList	*bufferList = nullptr;
+	AudioBufferList	*bufferList = null;
 	dataSize = 0;
-	result = AudioObjectGetPropertyDataSize(id, &property, 0, nullptr, &dataSize);
+	result = AudioObjectGetPropertyDataSize(id, &property, 0, null, &dataSize);
 	if (result != noErr || dataSize == 0) {
 		ATA_ERROR("system error (" << getErrorCode(result) << ") getting stream configuration info for device (" << _device << ").");
 		info.clear();
@@ -286,12 +286,12 @@ audio::orchestra::DeviceInfo audio::orchestra::api::Core::getDeviceInfo(uint32_t
 	}
 	// Allocate the AudioBufferList.
 	bufferList = (AudioBufferList *) malloc(dataSize);
-	if (bufferList == nullptr) {
+	if (bufferList == null) {
 		ATA_ERROR("memory error allocating AudioBufferList.");
 		info.clear();
 		return info;
 	}
-	result = AudioObjectGetPropertyData(id, &property, 0, nullptr, &dataSize, bufferList);
+	result = AudioObjectGetPropertyData(id, &property, 0, null, &dataSize, bufferList);
 	if (    result != noErr
 	     || dataSize == 0) {
 		free(bufferList);
@@ -302,7 +302,7 @@ audio::orchestra::DeviceInfo audio::orchestra::api::Core::getDeviceInfo(uint32_t
 	// Get channel information.
 	for (size_t iii=0; iii<bufferList->mNumberBuffers; ++iii) {
 		for (size_t jjj=0; jjj<bufferList->mBuffers[iii].mNumberChannels; ++jjj) {
-			info.channels.push_back(audio::channel_unknow);
+			info.channels.pushBack(audio::channel_unknow);
 		}
 	}
 	free(bufferList);
@@ -316,7 +316,7 @@ audio::orchestra::DeviceInfo audio::orchestra::api::Core::getDeviceInfo(uint32_t
 	// Determine the supported sample rates.
 	// ------------------------------------------------
 	property.mSelector = kAudioDevicePropertyAvailableNominalSampleRates;
-	result = AudioObjectGetPropertyDataSize(id, &property, 0, nullptr, &dataSize);
+	result = AudioObjectGetPropertyDataSize(id, &property, 0, null, &dataSize);
 	if (    result != kAudioHardwareNoError
 	     || dataSize == 0) {
 		ATA_ERROR("system error (" << getErrorCode(result) << ") getting sample rate info.");
@@ -325,7 +325,7 @@ audio::orchestra::DeviceInfo audio::orchestra::api::Core::getDeviceInfo(uint32_t
 	}
 	uint32_t nRanges = dataSize / sizeof(AudioValueRange);
 	AudioValueRange rangeList[ nRanges ];
-	result = AudioObjectGetPropertyData(id, &property, 0, nullptr, &dataSize, &rangeList);
+	result = AudioObjectGetPropertyData(id, &property, 0, null, &dataSize, &rangeList);
 	if (result != kAudioHardwareNoError) {
 		ATA_ERROR("system error (" << getErrorCode(result) << ") getting sample rates.");
 		info.clear();
@@ -344,7 +344,7 @@ audio::orchestra::DeviceInfo audio::orchestra::api::Core::getDeviceInfo(uint32_t
 	for (auto &it : audio::orchestra::genericSampleRate()) {
 		if (    it >= minimumRate
 			 && it <= maximumRate) {
-			info.sampleRates.push_back(it);
+			info.sampleRates.pushBack(it);
 		}
 	}
 	if (info.sampleRates.size() == 0) {
@@ -358,7 +358,7 @@ audio::orchestra::DeviceInfo audio::orchestra::api::Core::getDeviceInfo(uint32_t
 	// CoreAudio always uses 32-bit floating point data for PCM streams.
 	// Thus, any other "physical" formats supported by the device are of
 	// no interest to the client.
-	info.nativeFormats.push_back(audio::format_float);
+	info.nativeFormats.pushBack(audio::format_float);
 	// ------------------------------------------------
 	// Determine the default channel.
 	// ------------------------------------------------
@@ -386,10 +386,10 @@ OSStatus audio::orchestra::api::Core::callbackEvent(AudioDeviceID _inDevice,
 	audio::orchestra::api::Core* myClass = reinterpret_cast<audio::orchestra::api::Core*>(_userData);
 	audio::Time inputTime;
 	audio::Time outputTime;
-	if (_inInputTime != nullptr) {
+	if (_inInputTime != null) {
 		inputTime = audio::Time(_inInputTime->mHostTime/1000000000LL, _inInputTime->mHostTime%1000000000LL);
 	}
-	if (_inOutputTime != nullptr) {
+	if (_inOutputTime != null) {
 		outputTime = audio::Time(_inOutputTime->mHostTime/1000000000LL, _inOutputTime->mHostTime%1000000000LL);
 	}
 	if (myClass->callbackEvent(_inDevice, _inInputData, inputTime, _outOutputData, outputTime) == false) {
@@ -427,7 +427,7 @@ static OSStatus rateListener(AudioObjectID _inDevice,
 		kAudioObjectPropertyScopeGlobal,
 		kAudioObjectPropertyElementMaster
 	};
-	AudioObjectGetPropertyData(_inDevice, &property, 0, nullptr, &dataSize, rate);
+	AudioObjectGetPropertyData(_inDevice, &property, 0, null, &dataSize, rate);
 	return kAudioHardwareNoError;
 }
 
@@ -461,7 +461,7 @@ bool audio::orchestra::api::Core::open(uint32_t _device,
 	OSStatus result = AudioObjectGetPropertyData(kAudioObjectSystemObject,
 	                                             &property,
 	                                             0,
-	                                             nullptr,
+	                                             null,
 	                                             &dataSize,
 	                                             (void *) &deviceList);
 	if (result != noErr) {
@@ -481,7 +481,7 @@ bool audio::orchestra::api::Core::open(uint32_t _device,
 	AudioBufferList	*bufferList = nil;
 	dataSize = 0;
 	property.mSelector = kAudioDevicePropertyStreamConfiguration;
-	result = AudioObjectGetPropertyDataSize(id, &property, 0, nullptr, &dataSize);
+	result = AudioObjectGetPropertyDataSize(id, &property, 0, null, &dataSize);
 	if (    result != noErr
 	     || dataSize == 0) {
 		ATA_ERROR("system error (" << getErrorCode(result) << ") getting stream configuration info for device (" << _device << ").");
@@ -489,11 +489,11 @@ bool audio::orchestra::api::Core::open(uint32_t _device,
 	}
 	// Allocate the AudioBufferList.
 	bufferList = (AudioBufferList *) malloc(dataSize);
-	if (bufferList == nullptr) {
+	if (bufferList == null) {
 		ATA_ERROR("memory error allocating AudioBufferList.");
 		return false;
 	}
-	result = AudioObjectGetPropertyData(id, &property, 0, nullptr, &dataSize, bufferList);
+	result = AudioObjectGetPropertyData(id, &property, 0, null, &dataSize, bufferList);
 	if (    result != noErr
 	     || dataSize == 0) {
 		ATA_ERROR("system error (" << getErrorCode(result) << ") getting stream configuration for device (" << _device << ").");
@@ -569,7 +569,7 @@ bool audio::orchestra::api::Core::open(uint32_t _device,
 	AudioValueRange	bufferRange;
 	dataSize = sizeof(AudioValueRange);
 	property.mSelector = kAudioDevicePropertyBufferFrameSizeRange;
-	result = AudioObjectGetPropertyData(id, &property, 0, nullptr, &dataSize, &bufferRange);
+	result = AudioObjectGetPropertyData(id, &property, 0, null, &dataSize, &bufferRange);
 	if (result != noErr) {
 		ATA_ERROR("system error (" << getErrorCode(result) << ") getting buffer size range for device (" << _device << ").");
 		return false;
@@ -587,7 +587,7 @@ bool audio::orchestra::api::Core::open(uint32_t _device,
 	uint32_t theSize = (uint32_t) *_bufferSize;
 	dataSize = sizeof(uint32_t);
 	property.mSelector = kAudioDevicePropertyBufferFrameSize;
-	result = AudioObjectSetPropertyData(id, &property, 0, nullptr, dataSize, &theSize);
+	result = AudioObjectSetPropertyData(id, &property, 0, null, dataSize, &theSize);
 	if (result != noErr) {
 		ATA_ERROR("system error (" << getErrorCode(result) << ") setting the buffer size for device (" << _device << ").");
 		return false;
@@ -607,7 +607,7 @@ bool audio::orchestra::api::Core::open(uint32_t _device,
 	double nominalRate;
 	dataSize = sizeof(double);
 	property.mSelector = kAudioDevicePropertyNominalSampleRate;
-	result = AudioObjectGetPropertyData(id, &property, 0, nullptr, &dataSize, &nominalRate);
+	result = AudioObjectGetPropertyData(id, &property, 0, null, &dataSize, &nominalRate);
 	if (result != noErr) {
 		ATA_ERROR("system error (" << getErrorCode(result) << ") getting current sample rate.");
 		return false;
@@ -623,7 +623,7 @@ bool audio::orchestra::api::Core::open(uint32_t _device,
 			return false;
 		}
 		nominalRate = (double) _sampleRate;
-		result = AudioObjectSetPropertyData(id, &property, 0, nullptr, dataSize, &nominalRate);
+		result = AudioObjectSetPropertyData(id, &property, 0, null, dataSize, &nominalRate);
 		if (result != noErr) {
 			ATA_ERROR("system error (" << getErrorCode(result) << ") setting sample rate for device (" << _device << ").");
 			return false;
@@ -635,7 +635,7 @@ bool audio::orchestra::api::Core::open(uint32_t _device,
 			if (microCounter > 5000000) {
 				break;
 			}
-			usleep(5000);
+			ethread::sleepMilliSeconds((5));
 		}
 		// Remove the property listener.
 		AudioObjectRemovePropertyListener(id, &tmp, &rateListener, (void *) &reportedRate);
@@ -649,7 +649,7 @@ bool audio::orchestra::api::Core::open(uint32_t _device,
 	AudioStreamBasicDescription	description;
 	dataSize = sizeof(AudioStreamBasicDescription);
 	property.mSelector = kAudioStreamPropertyVirtualFormat;
-	result = AudioObjectGetPropertyData(id, &property, 0, nullptr, &dataSize, &description);
+	result = AudioObjectGetPropertyData(id, &property, 0, null, &dataSize, &description);
 	if (result != noErr) {
 		ATA_ERROR("system error (" << getErrorCode(result) << ") getting stream format for device (" << _device << ").");
 		return false;
@@ -667,7 +667,7 @@ bool audio::orchestra::api::Core::open(uint32_t _device,
 		updateFormat = true;
 	}
 	if (updateFormat) {
-		result = AudioObjectSetPropertyData(id, &property, 0, nullptr, dataSize, &description);
+		result = AudioObjectSetPropertyData(id, &property, 0, null, dataSize, &description);
 		if (result != noErr) {
 			ATA_ERROR("system error (" << getErrorCode(result) << ") setting sample rate or data format for device (" << _device << ").");
 			return false;
@@ -675,16 +675,16 @@ bool audio::orchestra::api::Core::open(uint32_t _device,
 	}
 	// Now check the physical format.
 	property.mSelector = kAudioStreamPropertyPhysicalFormat;
-	result = AudioObjectGetPropertyData(id, &property, 0, nullptr,	&dataSize, &description);
+	result = AudioObjectGetPropertyData(id, &property, 0, null,	&dataSize, &description);
 	if (result != noErr) {
 		ATA_ERROR("system error (" << getErrorCode(result) << ") getting stream physical format for device (" << _device << ").");
 		return false;
 	}
-	//std::cout << "Current physical stream format:" << std::endl;
-	//std::cout << "	 mBitsPerChan = " << description.mBitsPerChannel << std::endl;
-	//std::cout << "	 aligned high = " << (description.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (description.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
-	//std::cout << "	 bytesPerFrame = " << description.mBytesPerFrame << std::endl;
-	//std::cout << "	 sample rate = " << description.mSampleRate << std::endl;
+	//ATA_DEBUG("Current physical stream format:");
+	//ATA_DEBUG("	 mBitsPerChan = " << description.mBitsPerChannel);
+	//ATA_DEBUG("	 aligned high = " << (description.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (description.mFormatFlags & kAudioFormatFlagIsPacked));
+	//ATA_DEBUG("	 bytesPerFrame = " << description.mBytesPerFrame);
+	//ATA_DEBUG("	 sample rate = " << description.mSampleRate);
 	if (    description.mFormatID != kAudioFormatLinearPCM
 	     || description.mBitsPerChannel < 16) {
 		description.mFormatID = kAudioFormatLinearPCM;
@@ -692,19 +692,19 @@ bool audio::orchestra::api::Core::open(uint32_t _device,
 		AudioStreamBasicDescription	testDescription = description;
 		uint32_t formatFlags;
 		// We'll try higher bit rates first and then work our way down.
-		std::vector< std::pair<uint32_t, uint32_t>	> physicalFormats;
+		etk::Vector< etk::Pair<uint32_t, uint32_t>	> physicalFormats;
 		formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsFloat) & ~kLinearPCMFormatFlagIsSignedInteger;
-		physicalFormats.push_back(std::pair<float, uint32_t>(32, formatFlags));
+		physicalFormats.pushBack(etk::Pair<float, uint32_t>(32, formatFlags));
 		formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
-		physicalFormats.push_back(std::pair<float, uint32_t>(32, formatFlags));
-		physicalFormats.push_back(std::pair<float, uint32_t>(24, formatFlags));	 // 24-bit packed
+		physicalFormats.pushBack(etk::Pair<float, uint32_t>(32, formatFlags));
+		physicalFormats.pushBack(etk::Pair<float, uint32_t>(24, formatFlags));	 // 24-bit packed
 		formatFlags &= ~(kAudioFormatFlagIsPacked | kAudioFormatFlagIsAlignedHigh);
-		physicalFormats.push_back(std::pair<float, uint32_t>(24.2, formatFlags)); // 24-bit in 4 bytes, aligned low
+		physicalFormats.pushBack(etk::Pair<float, uint32_t>(24.2, formatFlags)); // 24-bit in 4 bytes, aligned low
 		formatFlags |= kAudioFormatFlagIsAlignedHigh;
-		physicalFormats.push_back(std::pair<float, uint32_t>(24.4, formatFlags)); // 24-bit in 4 bytes, aligned high
+		physicalFormats.pushBack(etk::Pair<float, uint32_t>(24.4, formatFlags)); // 24-bit in 4 bytes, aligned high
 		formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
-		physicalFormats.push_back(std::pair<float, uint32_t>(16, formatFlags));
-		physicalFormats.push_back(std::pair<float, uint32_t>(8, formatFlags));
+		physicalFormats.pushBack(etk::Pair<float, uint32_t>(16, formatFlags));
+		physicalFormats.pushBack(etk::Pair<float, uint32_t>(8, formatFlags));
 		bool setPhysicalFormat = false;
 		for(uint32_t i=0; i<physicalFormats.size(); i++) {
 			testDescription = description;
@@ -717,14 +717,14 @@ bool audio::orchestra::api::Core::open(uint32_t _device,
 				testDescription.mBytesPerFrame =	testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;
 			}
 			testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;
-			result = AudioObjectSetPropertyData(id, &property, 0, nullptr, dataSize, &testDescription);
+			result = AudioObjectSetPropertyData(id, &property, 0, null, dataSize, &testDescription);
 			if (result == noErr) {
 				setPhysicalFormat = true;
-				//std::cout << "Updated physical stream format:" << std::endl;
-				//std::cout << "	 mBitsPerChan = " << testDescription.mBitsPerChannel << std::endl;
-				//std::cout << "	 aligned high = " << (testDescription.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (testDescription.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
-				//std::cout << "	 bytesPerFrame = " << testDescription.mBytesPerFrame << std::endl;
-				//std::cout << "	 sample rate = " << testDescription.mSampleRate << std::endl;
+				//ATA_DEBUG("Updated physical stream format:");
+				//ATA_DEBUG("	 mBitsPerChan = " << testDescription.mBitsPerChannel);
+				//ATA_DEBUG("	 aligned high = " << (testDescription.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (testDescription.mFormatFlags & kAudioFormatFlagIsPacked));
+				//ATA_DEBUG("	 bytesPerFrame = " << testDescription.mBytesPerFrame);
+				//ATA_DEBUG("	 sample rate = " << testDescription.mSampleRate);
 				break;
 			}
 		}
@@ -738,7 +738,7 @@ bool audio::orchestra::api::Core::open(uint32_t _device,
 	dataSize = sizeof(uint32_t);
 	property.mSelector = kAudioDevicePropertyLatency;
 	if (AudioObjectHasProperty(id, &property) == true) {
-		result = AudioObjectGetPropertyData(id, &property, 0, nullptr, &dataSize, &latency);
+		result = AudioObjectGetPropertyData(id, &property, 0, null, &dataSize, &latency);
 		if (result == kAudioHardwareNoError) {
 			m_latency[ _mode ] = latency;
 		} else {
@@ -814,10 +814,10 @@ bool audio::orchestra::api::Core::open(uint32_t _device,
 			bufferBytes *= *_bufferSize;
 			if (m_deviceBuffer) {
 				free(m_deviceBuffer);
-				m_deviceBuffer = nullptr;
+				m_deviceBuffer = null;
 			}
 			m_deviceBuffer = (char *) calloc(bufferBytes, 1);
-			if (m_deviceBuffer == nullptr) {
+			if (m_deviceBuffer == null) {
 				ATA_ERROR("error allocating device buffer memory.");
 				goto error;
 			}
@@ -908,7 +908,7 @@ enum audio::orchestra::error audio::orchestra::api::Core::closeStream() {
 	m_userBuffer[1].clear();
 	if (m_deviceBuffer) {
 		free(m_deviceBuffer);
-		m_deviceBuffer = nullptr;
+		m_deviceBuffer = null;
 	}
 	m_mode = audio::orchestra::mode_unknow;
 	m_state = audio::orchestra::state::closed;
@@ -967,9 +967,9 @@ enum audio::orchestra::error audio::orchestra::api::Core::stopStream() {
 	if (    m_mode == audio::orchestra::mode_output
 	     || m_mode == audio::orchestra::mode_duplex) {
 		if (m_private->drainCounter == 0) {
-			std::unique_lock<std::mutex> lck(m_mutex);
+			ethread::UniqueLock lck(m_mutex);
 			m_private->drainCounter = 2;
-			m_private->condition.wait(lck);
+			m_private->m_semaphore.wait();
 		}
 		result = AudioDeviceStop(m_private->id[0], &audio::orchestra::api::Core::callbackEvent);
 		if (result != noErr) {
@@ -1036,10 +1036,10 @@ bool audio::orchestra::api::Core::callbackEvent(AudioDeviceID _deviceId,
 		m_state = audio::orchestra::state::stopping;
 		ATA_VERBOSE("Set state as stopping");
 		if (m_private->internalDrain == true) {
-			new std::thread(&audio::orchestra::api::Core::coreStopStream, this);
+			ETK_NEW(ethread::Thread, &audio::orchestra::api::Core::coreStopStream, this);
 		} else {
 			// external call to stopStream()
-			m_private->condition.notify_one();
+			m_private->m_semaphore.post();
 		}
 		return true;
 	}
@@ -1048,15 +1048,15 @@ bool audio::orchestra::api::Core::callbackEvent(AudioDeviceID _deviceId,
 	// draining stream or duplex mode AND the input/output devices are
 	// different AND this function is called for the input device.
 	if (m_private->drainCounter == 0 && (m_mode != audio::orchestra::mode_duplex || _deviceId == outputDevice)) {
-		std::vector<enum audio::orchestra::status> status;
+		etk::Vector<enum audio::orchestra::status> status;
 		if (    m_mode != audio::orchestra::mode_input
 		     && m_private->xrun[0] == true) {
-			status.push_back(audio::orchestra::status::underflow);
+			status.pushBack(audio::orchestra::status::underflow);
 			m_private->xrun[0] = false;
 		}
 		if (    m_mode != audio::orchestra::mode_output
 		     && m_private->xrun[1] == true) {
-			status.push_back(audio::orchestra::status::overflow);
+			status.pushBack(audio::orchestra::status::overflow);
 			m_private->xrun[1] = false;
 		}
 		int32_t cbReturnValue = m_callback(&m_userBuffer[1][0],

audio/orchestra/api/Core.hpp

@@ -20,7 +20,7 @@ namespace audio {
 				public:
 					Core();
 					virtual ~Core();
-					const std::string& getCurrentApi() {
+					const etk::String& getCurrentApi() {
 						return audio::orchestra::typeCoreOSX;
 					}
 					uint32_t getDeviceCount();

audio/orchestra/api/CoreIos.hpp

@@ -17,7 +17,7 @@ namespace audio {
 				public:
 					CoreIos();
 					virtual ~CoreIos();
-					const std::string& getCurrentApi() {
+					const etk::String& getCurrentApi() {
 						return audio::orchestra::typeCoreIOS;
 					}
 					uint32_t getDeviceCount();
@@ -32,7 +32,7 @@ namespace audio {
 					// will most likely produce highly undesireable results!
 					void callbackEvent();
 				private:
-					std::vector<audio::orchestra::DeviceInfo> m_devices;
+					etk::Vector<audio::orchestra::DeviceInfo> m_devices;
 					void saveDeviceInfo();
 					bool open(uint32_t _device,
 					          audio::orchestra::mode _mode,
@@ -48,6 +48,8 @@ namespace audio {
 					                   const audio::Time& _time);
 				public:
 					ememory::SharedPtr<CoreIosPrivate> m_private;
+					uint32_t getDefaultInputDevice();
+					uint32_t getDefaultOutputDevice();
 			};
 		}
 	}

audio/orchestra/api/CoreIos.mm

@@ -10,15 +10,17 @@
 #import <Foundation/Foundation.h>
 #import <AudioToolbox/AudioToolbox.h>
 
-#include <unistd.h>
-#include <audio/orchestra/Interface.h>
-#include <audio/orchestra/debug.h>
-#include <limits.h>
-#include <audio/orchestra/api/CoreIos.h>
+
+#include <audio/orchestra/Interface.hpp>
+#include <audio/orchestra/debug.hpp>
+extern "C" {
+	#include <limits.h>
+}
+#include <audio/orchestra/api/CoreIos.hpp>
 
 ememory::SharedPtr<audio::orchestra::Api> audio::orchestra::api::CoreIos::create() {
 	ATA_INFO("Create CoreIos device ... ");
-	return ememory::SharedPtr<audio::orchestra::api::CoreIos>(new audio::orchestra::api::CoreIos());
+	return ememory::SharedPtr<audio::orchestra::api::CoreIos>(ETK_NEW(audio::orchestra::api::CoreIos));
 }
 
 #define kOutputBus 0
@@ -37,53 +39,75 @@ namespace audio {
 
 
 
-audio::orchestra::api::CoreIos::CoreIos(void) :
-  m_private(new audio::orchestra::api::CoreIosPrivate()) {
+audio::orchestra::api::CoreIos::CoreIos() :
+  m_private(ETK_NEW(audio::orchestra::api::CoreIosPrivate)) {
 	ATA_INFO("new CoreIos");
 	int32_t deviceCount = 2;
 	ATA_ERROR("Get count devices : " << 2);
 	audio::orchestra::DeviceInfo tmp;
 	// Add default output format :
 	tmp.name = "speaker";
+	tmp.input = false;
 	tmp.sampleRates.push_back(48000);
 	tmp.channels.push_back(audio::channel_frontRight);
 	tmp.channels.push_back(audio::channel_frontLeft);
-	tmp.isDefault = true;
 	tmp.nativeFormats.push_back(audio::format_int16);
+	tmp.isDefault = true;
+	tmp.isCorrect = true;
 	m_devices.push_back(tmp);
 	// add default input format:
 	tmp.name = "microphone";
+	tmp.input = true;
 	tmp.sampleRates.push_back(48000);
 	tmp.channels.push_back(audio::channel_frontRight);
 	tmp.channels.push_back(audio::channel_frontLeft);
-	tmp.isDefault = true;
 	tmp.nativeFormats.push_back(audio::format_int16);
+	tmp.isDefault = true;
+	tmp.isCorrect = true;
 	m_devices.push_back(tmp);
 	ATA_INFO("Create CoreIOs interface (end)");
 }
 
-audio::orchestra::api::CoreIos::~CoreIos(void) {
+uint32_t audio::orchestra::api::CoreIos::getDefaultInputDevice() {
+	// Should be implemented in subclasses if possible.
+	return 1;
+}
+
+uint32_t audio::orchestra::api::CoreIos::getDefaultOutputDevice() {
+	// Should be implemented in subclasses if possible.
+	return 0;
+}
+
+audio::orchestra::api::CoreIos::~CoreIos() {
 	ATA_INFO("Destroy CoreIOs interface");
 	AudioUnitUninitialize(m_private->audioUnit);
 }
 
-uint32_t audio::orchestra::api::CoreIos::getDeviceCount(void) {
+uint32_t audio::orchestra::api::CoreIos::getDeviceCount() {
 	//ATA_INFO("Get device count:"<< m_devices.size());
 	return m_devices.size();
 }
 
 audio::orchestra::DeviceInfo audio::orchestra::api::CoreIos::getDeviceInfo(uint32_t _device) {
 	//ATA_INFO("Get device info ...");
+	if (_device >= m_devices.size()) {
+		audio::orchestra::DeviceInfo tmp;
+		tmp.sampleRates.push_back(0);
+		tmp.channels.push_back(audio::channel_frontCenter);
+		tmp.isDefault = false;
+		tmp.nativeFormats.push_back(audio::format_int8);
+		return tmp;
+	}
 	return m_devices[_device];
 }
 
-enum audio::orchestra::error audio::orchestra::api::CoreIos::closeStream(void) {
+enum audio::orchestra::error audio::orchestra::api::CoreIos::closeStream() {
 	ATA_INFO("Close Stream");
 	// Can not close the stream now...
 	return audio::orchestra::error_none;
 }
 
-enum audio::orchestra::error audio::orchestra::api::CoreIos::startStream(void) {
+enum audio::orchestra::error audio::orchestra::api::CoreIos::startStream() {
 	ATA_INFO("Start Stream");
 	// TODO : Check return ...
 	audio::orchestra::Api::startStream();
@@ -92,14 +116,14 @@ enum audio::orchestra::error audio::orchestra::api::CoreIos::startStream(void) {
 	return audio::orchestra::error_none;
 }
 
-enum audio::orchestra::error audio::orchestra::api::CoreIos::stopStream(void) {
+enum audio::orchestra::error audio::orchestra::api::CoreIos::stopStream() {
 	ATA_INFO("Stop stream");
 	OSStatus status = AudioOutputUnitStop(m_private->audioUnit);
 	// Can not close the stream now...
 	return audio::orchestra::error_none;
 }
 
-enum audio::orchestra::error audio::orchestra::api::CoreIos::abortStream(void) {
+enum audio::orchestra::error audio::orchestra::api::CoreIos::abortStream() {
 	ATA_INFO("Abort Stream");
 	OSStatus status = AudioOutputUnitStop(m_private->audioUnit);
 	// Can not close the stream now...
@@ -110,19 +134,34 @@ void audio::orchestra::api::CoreIos::callBackEvent(void* _data,
                                                    int32_t _nbChunk,
                                                    const audio::Time& _time) {
 	int32_t doStopStream = 0;
-	std::vector<enum audio::orchestra::status> status;
-	if (m_doConvertBuffer[modeToIdTable(audio::orchestra::mode_output)] == true) {
-		doStopStream = m_callback(nullptr,
-		                          audio::Time(),
-		                          &m_userBuffer[modeToIdTable(audio::orchestra::mode_output)][0],
-		                          _time,
-		                          _nbChunk,
-		                          status);
-		convertBuffer((char*)_data, &m_userBuffer[modeToIdTable(audio::orchestra::mode_output)][0], m_convertInfo[modeToIdTable(audio::orchestra::mode_output)]);
-	} else {
+	etk::Vector<enum audio::orchestra::status> status;
+	if (    m_mode == audio::orchestra::mode_output
+	     || m_mode == audio::orchestra::mode_duplex) {
+		if (m_doConvertBuffer[modeToIdTable(audio::orchestra::mode_output)] == true) {
+			ATA_INFO("get output DATA : " << uint64_t(&m_userBuffer[modeToIdTable(audio::orchestra::mode_output)][0]));
+			doStopStream = m_callback(null,
+			                          audio::Time(),
+			                          &m_userBuffer[modeToIdTable(audio::orchestra::mode_output)][0],
+			                          _time,
+			                          _nbChunk,
+			                          status);
+			convertBuffer((char*)_data, &m_userBuffer[modeToIdTable(audio::orchestra::mode_output)][0], m_convertInfo[modeToIdTable(audio::orchestra::mode_output)]);
+		} else {
+			ATA_INFO("have output DATA : " << uint64_t(_data));
+			doStopStream = m_callback(null,
+			                          _time,
+			                          _data,
+			                          audio::Time(),
+			                          _nbChunk,
+			                          status);
+		}
+	}
+	if (    m_mode == audio::orchestra::mode_input
+	     || m_mode == audio::orchestra::mode_duplex) {
+		ATA_INFO("have input DATA : " << uint64_t(_data));
 		doStopStream = m_callback(_data,
 		                          _time,
-		                          nullptr,
+		                          null,
 		                          audio::Time(),
 		                          _nbChunk,
 		                          status);
@@ -141,12 +180,12 @@ static OSStatus playbackCallback(void *_userData,
                                  uint32_t _inBusNumber,
                                  uint32_t _inNumberFrames,
                                  AudioBufferList* _ioData) {
-	if (_userData == nullptr) {
-		ATA_ERROR("callback event ... nullptr pointer");
+	if (_userData == null) {
+		ATA_ERROR("callback event ... null pointer");
 		return -1;
 	}
 	audio::Time tmpTimeime;
-	if (_inTime != nullptr) {
+	if (_inTime != null) {
 		tmpTimeime = audio::Time(_inTime->mHostTime/1000000000LL, _inTime->mHostTime%1000000000LL);
 	}
 	audio::orchestra::api::CoreIos* myClass = static_cast<audio::orchestra::api::CoreIos*>(_userData);
@@ -154,7 +193,7 @@ static OSStatus playbackCallback(void *_userData,
 	for (int32_t iii=0; iii < _ioData->mNumberBuffers; iii++) {
 		AudioBuffer buffer = _ioData->mBuffers[iii];
 		int32_t numberFrame = buffer.mDataByteSize/2/*stereo*/ /sizeof(int16_t);
-		ATA_VERBOSE("request data size: " << numberFrame << " busNumber=" << _inBusNumber);
+		ATA_INFO("request data size: " << numberFrame << " busNumber=" << _inBusNumber);
 		myClass->callBackEvent(buffer.mData, numberFrame, tmpTimeime);
 	}
 	return noErr;
@@ -175,7 +214,8 @@ bool audio::orchestra::api::CoreIos::open(uint32_t _device,
 		return false;
 	}
 	bool ret = true;
-	
+	// TODO : This is a bad ack ....
+	m_mode = audio::orchestra::mode_output;
 	// configure Airtaudio internal configuration:
 	m_userFormat = _format;
 	m_nUserChannels[modeToIdTable(_mode)] = _channels;
@@ -227,7 +267,7 @@ bool audio::orchestra::api::CoreIos::open(uint32_t _device,
 	desc.componentManufacturer = kAudioUnitManufacturer_Apple;
 	
 	// Get component
-	AudioComponent inputComponent = AudioComponentFindNext(nullptr, &desc);
+	AudioComponent inputComponent = AudioComponentFindNext(null, &desc);
 	
 	// Get audio units
 	status = AudioComponentInstanceNew(inputComponent, &m_private->audioUnit);
@@ -238,11 +278,11 @@ bool audio::orchestra::api::CoreIos::open(uint32_t _device,
 	uint32_t flag = 1;
 	// Enable IO for playback
 	status = AudioUnitSetProperty(m_private->audioUnit, 
-								  kAudioOutputUnitProperty_EnableIO, 
-								  kAudioUnitScope_Output, 
-								  kOutputBus,
-								  &flag, 
-								  sizeof(flag));
+	                              kAudioOutputUnitProperty_EnableIO, 
+	                              kAudioUnitScope_Output, 
+	                              kOutputBus,
+	                              &flag, 
+	                              sizeof(flag));
 	if (status != 0) {
 		ATA_ERROR("can not request audio autorisation...");
 	}
@@ -260,11 +300,11 @@ bool audio::orchestra::api::CoreIos::open(uint32_t _device,
 	audioFormat.mReserved = 0;
 	// Apply format
 	status = AudioUnitSetProperty(m_private->audioUnit, 
-								  kAudioUnitProperty_StreamFormat, 
-								  kAudioUnitScope_Input, 
-								  kOutputBus, 
-								  &audioFormat, 
-								  sizeof(audioFormat));
+	                              kAudioUnitProperty_StreamFormat, 
+	                              kAudioUnitScope_Input, 
+	                              kOutputBus, 
+	                              &audioFormat, 
+	                              sizeof(audioFormat));
 	if (status != 0) {
 		ATA_ERROR("can not set stream properties...");
 	}
@@ -275,11 +315,11 @@ bool audio::orchestra::api::CoreIos::open(uint32_t _device,
 	callbackStruct.inputProc = &playbackCallback;
 	callbackStruct.inputProcRefCon = this;
 	status = AudioUnitSetProperty(m_private->audioUnit, 
-								  kAudioUnitProperty_SetRenderCallback, 
-								  kAudioUnitScope_Global, 
-								  kOutputBus,
-								  &callbackStruct, 
-								  sizeof(callbackStruct));
+	                              kAudioUnitProperty_SetRenderCallback, 
+	                              kAudioUnitScope_Global, 
+	                              kOutputBus,
+	                              &callbackStruct, 
+	                              sizeof(callbackStruct));
 	if (status != 0) {
 		ATA_ERROR("can not set Callback...");
 	}

audio/orchestra/api/Ds.cpp

@@ -7,13 +7,13 @@
 
 // Windows DirectSound API
 #if defined(ORCHESTRA_BUILD_DS)
-#include <audio/orchestra/Interface.h>
-#include <audio/orchestra/debug.h>
-#include <ethread/tools.h>
-#include <audio/orchestra/api/Ds.h>
+#include <audio/orchestra/Interface.hpp>
+#include <audio/orchestra/debug.hpp>
+#include <ethread/tools.hpp>
+#include <audio/orchestra/api/Ds.hpp>
 
 ememory::SharedPtr<audio::orchestra::Api> audio::orchestra::api::Ds::create() {
-	return ememory::SharedPtr<audio::orchestra::api::Ds>(new audio::orchestra::api::Ds());
+	return ememory::SharedPtr<audio::orchestra::api::Ds>(ETK_NEW(audio::orchestra::api::Ds));
 }
 
 
@@ -23,11 +23,10 @@ ememory::SharedPtr<audio::orchestra::Api> audio::orchestra::api::Ds::create() {
 // - Auto-call CoInitialize for DSOUND and ASIO platforms.
 // Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
 // Changed device query structure for RtAudio 4.0.7, January 2010
-
-#include <dsound.h>
-#include <assert.h>
-#include <algorithm>
-
+extern "C" {
+	#include <dsound.h>
+	#include <assert.h>
+}
 #if defined(__MINGW32__)
 	// missing from latest mingw winapi
 #define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */
@@ -59,7 +58,7 @@ class DsDevice {
 	public:
 		LPGUID id;
 		bool input;
-		std::string name;
+		etk::String name;
 		DsDevice() :
 		  id(0),
 		  input(false) {
@@ -72,7 +71,7 @@ namespace audio {
 		namespace api {
 			class DsPrivate {
 				public:
-					ememory::SharedPtr<std::thread> thread;
+					ememory::SharedPtr<ethread::Thread> thread;
 					bool threadRunning;
 					uint32_t drainCounter; // Tracks callback counts when draining
 					bool internalDrain; // Indicates if stop is initiated from callback or not.
@@ -83,7 +82,7 @@ namespace audio {
 					DWORD dsBufferSize[2];
 					DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.
 					HANDLE condition;
-					std::vector<DsDevice> dsDevices;
+					etk::Vector<DsDevice> dsDevices;
 					DsPrivate() :
 					  threadRunning(false),
 					  drainCounter(0),
@@ -107,15 +106,15 @@ static const char* getErrorString(int32_t _code);
 
 struct DsProbeData {
 	bool isInput;
-	std::vector<DsDevice>* dsDevices;
+	etk::Vector<DsDevice>* dsDevices;
 };
 
 audio::orchestra::api::Ds::Ds() :
-  m_private(new audio::orchestra::api::DsPrivate()) {
+  m_private(ETK_NEW(audio::orchestra::api::DsPrivate)) {
 	// Dsound will run both-threaded. If CoInitialize fails, then just
 	// accept whatever the mainline chose for a threading model.
 	m_coInitialized = false;
-	HRESULT hr = CoInitialize(nullptr);
+	HRESULT hr = CoInitialize(null);
 	if (!FAILED(hr)) {
 		m_coInitialized = true;
 	}
@@ -132,13 +131,13 @@ audio::orchestra::api::Ds::~Ds() {
 
 
 #include "tchar.h"
-static std::string convertTChar(LPCTSTR _name) {
+static etk::String convertTChar(LPCTSTR _name) {
 #if defined(UNICODE) || defined(_UNICODE)
-	int32_t length = WideCharToMultiByte(CP_UTF8, 0, _name, -1, nullptr, 0, nullptr, nullptr);
-	std::string s(length-1, '\0');
-	WideCharToMultiByte(CP_UTF8, 0, _name, -1, &s[0], length, nullptr, nullptr);
+	int32_t length = WideCharToMultiByte(CP_UTF8, 0, _name, -1, null, 0, null, null);
+	etk::String s(length-1, '\0');
+	WideCharToMultiByte(CP_UTF8, 0, _name, -1, &s[0], length, null, null);
 #else
-	std::string s(_name);
+	etk::String s(_name);
 #endif
 	return s;
 }
@@ -148,13 +147,13 @@ static BOOL CALLBACK deviceQueryCallback(LPGUID _lpguid,
                                          LPCTSTR _module,
                                          LPVOID _lpContext) {
 	struct DsProbeData& probeInfo = *(struct DsProbeData*) _lpContext;
-	std::vector<DsDevice>& dsDevices = *probeInfo.dsDevices;
+	etk::Vector<DsDevice>& dsDevices = *probeInfo.dsDevices;
 	HRESULT hr;
 	bool validDevice = false;
 	if (probeInfo.isInput == true) {
 		DSCCAPS caps;
 		LPDIRECTSOUNDCAPTURE object;
-		hr = DirectSoundCaptureCreate(_lpguid, &object, nullptr);
+		hr = DirectSoundCaptureCreate(_lpguid, &object, null);
 		if (hr != DS_OK) {
 			return TRUE;
 		}
@@ -169,7 +168,7 @@ static BOOL CALLBACK deviceQueryCallback(LPGUID _lpguid,
 	} else {
 		DSCAPS caps;
 		LPDIRECTSOUND object;
-		hr = DirectSoundCreate(_lpguid, &object, nullptr);
+		hr = DirectSoundCreate(_lpguid, &object, null);
 		if (hr != DS_OK) {
 			return TRUE;
 		}
@@ -187,16 +186,16 @@ static BOOL CALLBACK deviceQueryCallback(LPGUID _lpguid,
 		return TRUE;
 	}
 	// If good device, then save its name and guid.
-	std::string name = convertTChar(_description);
+	etk::String name = convertTChar(_description);
 	//if (name == "Primary Sound Driver" || name == "Primary Sound Capture Driver")
-	if (_lpguid == nullptr) {
+	if (_lpguid == null) {
 		name = "Default Device";
 	}
 	DsDevice device;
 	device.name = name;
 	device.input = probeInfo.isInput;
 	device.id = _lpguid;
-	dsDevices.push_back(device);
+	dsDevices.pushBack(device);
 	return TRUE;
 }
 
@@ -234,7 +233,7 @@ audio::orchestra::DeviceInfo audio::orchestra::api::Ds::getDeviceInfo(uint32_t _
 		info.input = true;
 		LPDIRECTSOUND output;
 		DSCAPS outCaps;
-		result = DirectSoundCreate(m_private->dsDevices[_device].id, &output, nullptr);
+		result = DirectSoundCreate(m_private->dsDevices[_device].id, &output, null);
 		if (FAILED(result)) {
 			ATA_ERROR(getErrorString(result) << ": opening output device (" << m_private->dsDevices[_device].name << ")!");
 			info.clear();
@@ -250,24 +249,24 @@ audio::orchestra::DeviceInfo audio::orchestra::api::Ds::getDeviceInfo(uint32_t _
 		}
 		// Get output channel information.
 		if (outCaps.dwFlags & DSCAPS_PRIMARYSTEREO) {
-			info.channels.push_back(audio::channel_unknow);
-			info.channels.push_back(audio::channel_unknow);
+			info.channels.pushBack(audio::channel_unknow);
+			info.channels.pushBack(audio::channel_unknow);
 		} else {
-			info.channels.push_back(audio::channel_unknow);
+			info.channels.pushBack(audio::channel_unknow);
 		}
 		// Get sample rate information.
 		for (auto &it : audio::orchestra::genericSampleRate()) {
 			if (    it >= outCaps.dwMinSecondarySampleRate
 			     && it <= outCaps.dwMaxSecondarySampleRate) {
-				info.sampleRates.push_back(it);
+				info.sampleRates.pushBack(it);
 			}
 		}
 		// Get format information.
 		if (outCaps.dwFlags & DSCAPS_PRIMARY16BIT) {
-			info.nativeFormats.push_back(audio::format_int16);
+			info.nativeFormats.pushBack(audio::format_int16);
 		}
 		if (outCaps.dwFlags & DSCAPS_PRIMARY8BIT) {
-			info.nativeFormats.push_back(audio::format_int8);
+			info.nativeFormats.pushBack(audio::format_int8);
 		}
 		output->Release();
 		info.name = m_private->dsDevices[_device].name;
@@ -275,7 +274,7 @@ audio::orchestra::DeviceInfo audio::orchestra::api::Ds::getDeviceInfo(uint32_t _
 		return info;
 	} else {
 		LPDIRECTSOUNDCAPTURE input;
-		result = DirectSoundCaptureCreate(m_private->dsDevices[_device].id, &input, nullptr);
+		result = DirectSoundCaptureCreate(m_private->dsDevices[_device].id, &input, null);
 		if (FAILED(result)) {
 			ATA_ERROR(getErrorString(result) << ": opening input device (" << m_private->dsDevices[_device].name << ")!");
 			info.clear();
@@ -292,67 +291,67 @@ audio::orchestra::DeviceInfo audio::orchestra::api::Ds::getDeviceInfo(uint32_t _
 		}
 		// Get input channel information.
 		for (int32_t iii=0; iii<inCaps.dwChannels; ++iii) {
-			info.channels.push_back(audio::channel_unknow);
+			info.channels.pushBack(audio::channel_unknow);
 		}
 		// Get sample rate and format information.
-		std::vector<uint32_t> rates;
+		etk::Vector<uint32_t> rates;
 		if (inCaps.dwChannels >= 2) {
 			if (    (inCaps.dwFormats & WAVE_FORMAT_1S16)
 			     || (inCaps.dwFormats & WAVE_FORMAT_2S16)
 			     || (inCaps.dwFormats & WAVE_FORMAT_4S16)
 			     || (inCaps.dwFormats & WAVE_FORMAT_96S16) ) {
-				info.nativeFormats.push_back(audio::format_int16);
+				info.nativeFormats.pushBack(audio::format_int16);
 			}
 			if (    (inCaps.dwFormats & WAVE_FORMAT_1S08)
 			     || (inCaps.dwFormats & WAVE_FORMAT_2S08)
 			     || (inCaps.dwFormats & WAVE_FORMAT_4S08)
 			     || (inCaps.dwFormats & WAVE_FORMAT_96S08) ) {
-				info.nativeFormats.push_back(audio::format_int8);
+				info.nativeFormats.pushBack(audio::format_int8);
 			}
 			if (    (inCaps.dwFormats & WAVE_FORMAT_1S16)
 			     || (inCaps.dwFormats & WAVE_FORMAT_1S08) ){
-				rates.push_back(11025);
+				rates.pushBack(11025);
 			}
 			if (    (inCaps.dwFormats & WAVE_FORMAT_2S16)
 			     || (inCaps.dwFormats & WAVE_FORMAT_2S08) ){
-				rates.push_back(22050);
+				rates.pushBack(22050);
 			}
 			if (    (inCaps.dwFormats & WAVE_FORMAT_4S16)
 			     || (inCaps.dwFormats & WAVE_FORMAT_4S08) ){
-				rates.push_back(44100);
+				rates.pushBack(44100);
 			}
 			if (    (inCaps.dwFormats & WAVE_FORMAT_96S16)
 			     || (inCaps.dwFormats & WAVE_FORMAT_96S08) ){
-				rates.push_back(96000);
+				rates.pushBack(96000);
 			}
 		} else if (inCaps.dwChannels == 1) {
 			if (    (inCaps.dwFormats & WAVE_FORMAT_1M16)
 			     || (inCaps.dwFormats & WAVE_FORMAT_2M16)
 			     || (inCaps.dwFormats & WAVE_FORMAT_4M16)
 			     || (inCaps.dwFormats & WAVE_FORMAT_96M16) ) {
-				info.nativeFormats.push_back(audio::format_int16);
+				info.nativeFormats.pushBack(audio::format_int16);
 			}
 			if (    (inCaps.dwFormats & WAVE_FORMAT_1M08)
 			     || (inCaps.dwFormats & WAVE_FORMAT_2M08)
 			     || (inCaps.dwFormats & WAVE_FORMAT_4M08)
 			     || (inCaps.dwFormats & WAVE_FORMAT_96M08) ) {
-				info.nativeFormats.push_back(audio::format_int8);
+				info.nativeFormats.pushBack(audio::format_int8);
 			}
 			if (    (inCaps.dwFormats & WAVE_FORMAT_1M16)
 			     || (inCaps.dwFormats & WAVE_FORMAT_1M08) ){
-				rates.push_back(11025);
+				rates.pushBack(11025);
 			}
 			if (    (inCaps.dwFormats & WAVE_FORMAT_2M16)
 			     || (inCaps.dwFormats & WAVE_FORMAT_2M08) ){
-				rates.push_back(22050);
+				rates.pushBack(22050);
 			}
 			if (    (inCaps.dwFormats & WAVE_FORMAT_4M16)
 			     || (inCaps.dwFormats & WAVE_FORMAT_4M08) ){
-				rates.push_back(44100);
+				rates.pushBack(44100);
 			}
 			if (    (inCaps.dwFormats & WAVE_FORMAT_96M16)
 			     || (inCaps.dwFormats & WAVE_FORMAT_96M08) ){
-				rates.push_back(96000);
+				rates.pushBack(96000);
 			}
 		} else {
 			// technically, this would be an error
@@ -374,10 +373,10 @@ audio::orchestra::DeviceInfo audio::orchestra::api::Ds::getDeviceInfo(uint32_t _
 				}
 			}
 			if (found == false) {
-				info.sampleRates.push_back(rates[i]);
+				info.sampleRates.pushBack(rates[i]);
 			}
 		}
-		std::sort(info.sampleRates.begin(), info.sampleRates.end());
+		info.sampleRates.sort(0, info.sampleRates.size(), [](const uint32_t& _left, const uint32_t& _right) { return _left < _right;});
 		// Copy name and return.
 		info.name = m_private->dsDevices[_device].name;
 		info.isCorrect = true;
@@ -453,7 +452,7 @@ bool audio::orchestra::api::Ds::open(uint32_t _device,
 	HRESULT result;
 	if (_mode == audio::orchestra::mode_output) {
 		LPDIRECTSOUND output;
-		result = DirectSoundCreate(m_private->dsDevices[_device].id, &output, nullptr);
+		result = DirectSoundCreate(m_private->dsDevices[_device].id, &output, null);
 		if (FAILED(result)) {
 			ATA_ERROR(getErrorString(result) << ": opening output device (" << m_private->dsDevices[_device].name << ")!");
 			return false;
@@ -510,7 +509,7 @@ bool audio::orchestra::api::Ds::open(uint32_t _device,
 		bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
 		// Obtain the primary buffer
 		LPDIRECTSOUNDBUFFER buffer;
-		result = output->CreateSoundBuffer(&bufferDescription, &buffer, nullptr);
+		result = output->CreateSoundBuffer(&bufferDescription, &buffer, null);
 		if (FAILED(result)) {
 			output->Release();
 			ATA_ERROR(getErrorString(result) << ": accessing primary buffer (" << m_private->dsDevices[_device].name << ")!");
@@ -534,13 +533,13 @@ bool audio::orchestra::api::Ds::open(uint32_t _device,
 		bufferDescription.lpwfxFormat = &waveFormat;
 		// Try to create the secondary DS buffer.	If that doesn't work,
 		// try to use software mixing.	Otherwise, there's a problem.
-		result = output->CreateSoundBuffer(&bufferDescription, &buffer, nullptr);
+		result = output->CreateSoundBuffer(&bufferDescription, &buffer, null);
 		if (FAILED(result)) {
 			bufferDescription.dwFlags = (   DSBCAPS_STICKYFOCUS
 			                              | DSBCAPS_GLOBALFOCUS
 			                              | DSBCAPS_GETCURRENTPOSITION2
 			                              | DSBCAPS_LOCSOFTWARE);	// Force software mixing
-			result = output->CreateSoundBuffer(&bufferDescription, &buffer, nullptr);
+			result = output->CreateSoundBuffer(&bufferDescription, &buffer, null);
 			if (FAILED(result)) {
 				output->Release();
 				ATA_ERROR(getErrorString(result) << ": creating secondary buffer (" << m_private->dsDevices[_device].name << ")!");
@@ -561,7 +560,7 @@ bool audio::orchestra::api::Ds::open(uint32_t _device,
 		// Lock the DS buffer
 		LPVOID audioPtr;
 		DWORD dataLen;
-		result = buffer->Lock(0, dsBufferSize, &audioPtr, &dataLen, nullptr, nullptr, 0);
+		result = buffer->Lock(0, dsBufferSize, &audioPtr, &dataLen, null, null, 0);
 		if (FAILED(result)) {
 			output->Release();
 			buffer->Release();
@@ -571,7 +570,7 @@ bool audio::orchestra::api::Ds::open(uint32_t _device,
 		// Zero the DS buffer
 		ZeroMemory(audioPtr, dataLen);
 		// Unlock the DS buffer
-		result = buffer->Unlock(audioPtr, dataLen, nullptr, 0);
+		result = buffer->Unlock(audioPtr, dataLen, null, 0);
 		if (FAILED(result)) {
 			output->Release();
 			buffer->Release();
@@ -583,7 +582,7 @@ bool audio::orchestra::api::Ds::open(uint32_t _device,
 	}
 	if (_mode == audio::orchestra::mode_input) {
 		LPDIRECTSOUNDCAPTURE input;
-		result = DirectSoundCaptureCreate(m_private->dsDevices[_device].id, &input, nullptr);
+		result = DirectSoundCaptureCreate(m_private->dsDevices[_device].id, &input, null);
 		if (FAILED(result)) {
 			ATA_ERROR(getErrorString(result) << ": opening input device (" << m_private->dsDevices[_device].name << ")!");
 			return false;
@@ -643,7 +642,7 @@ bool audio::orchestra::api::Ds::open(uint32_t _device,
 		bufferDescription.lpwfxFormat = &waveFormat;
 		// Create the capture buffer.
 		LPDIRECTSOUNDCAPTUREBUFFER buffer;
-		result = input->CreateCaptureBuffer(&bufferDescription, &buffer, nullptr);
+		result = input->CreateCaptureBuffer(&bufferDescription, &buffer, null);
 		if (FAILED(result)) {
 			input->Release();
 			ATA_ERROR(getErrorString(result) << ": creating input buffer (" << m_private->dsDevices[_device].name << ")!");
@@ -667,7 +666,7 @@ bool audio::orchestra::api::Ds::open(uint32_t _device,
 		// Lock the capture buffer
 		LPVOID audioPtr;
 		DWORD dataLen;
-		result = buffer->Lock(0, dsBufferSize, &audioPtr, &dataLen, nullptr, nullptr, 0);
+		result = buffer->Lock(0, dsBufferSize, &audioPtr, &dataLen, null, null, 0);
 		if (FAILED(result)) {
 			input->Release();
 			buffer->Release();
@@ -677,7 +676,7 @@ bool audio::orchestra::api::Ds::open(uint32_t _device,
 		// Zero the buffer
 		ZeroMemory(audioPtr, dataLen);
 		// Unlock the buffer
-		result = buffer->Unlock(audioPtr, dataLen, nullptr, 0);
+		result = buffer->Unlock(audioPtr, dataLen, null, 0);
 		if (FAILED(result)) {
 			input->Release();
 			buffer->Release();
@@ -729,17 +728,17 @@ bool audio::orchestra::api::Ds::open(uint32_t _device,
 				free(m_deviceBuffer);
 			}
 			m_deviceBuffer = (char *) calloc(bufferBytes, 1);
-			if (m_deviceBuffer == nullptr) {
+			if (m_deviceBuffer == null) {
 				ATA_ERROR("error allocating device buffer memory.");
 				goto error;
 			}
 		}
 	}
 	// Create a manual-reset event.
-	m_private->condition = CreateEvent(nullptr,  // no security
+	m_private->condition = CreateEvent(null,  // no security
 	                                TRUE,  // manual-reset
 	                                FALSE, // non-signaled initially
-	                                nullptr); // unnamed
+	                                null); // unnamed
 	m_private->id[modeToIdTable(_mode)] = ohandle;
 	m_private->buffer[modeToIdTable(_mode)] = bhandle;
 	m_private->dsBufferSize[modeToIdTable(_mode)] = dsBufferSize;
@@ -762,9 +761,9 @@ bool audio::orchestra::api::Ds::open(uint32_t _device,
 	// Setup the callback thread.
 	if (m_private->threadRunning == false) {
 		m_private->threadRunning = true;
-		ememory::SharedPtr<std::thread> tmpThread(new std::thread(&audio::orchestra::api::Ds::dsCallbackEvent, this));
-		m_private->thread =	std::move(tmpThread);
-		if (m_private->thread == nullptr) {
+		ememory::SharedPtr<ethread::Thread> tmpThread(ETK_NEW(ethread::Thread, [=](){audio::orchestra::api::Ds::dsCallbackEvent();});
+		m_private->thread =	etk::move(tmpThread);
+		if (m_private->thread == null) {
 			ATA_ERROR("error creating callback thread!");
 			goto error;
 		}
@@ -774,7 +773,7 @@ bool audio::orchestra::api::Ds::open(uint32_t _device,
 	return true;
 error:
 	if (m_private->buffer[0]) {
-		// the object pointer can be nullptr and valid
+		// the object pointer can be null and valid
 		LPDIRECTSOUND object = (LPDIRECTSOUND) m_private->id[0];
 		LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) m_private->buffer[0];
 		if (buffer) {
@@ -785,7 +784,7 @@ error:
 	if (m_private->buffer[1]) {
 		LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) m_private->id[1];
 		LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) m_private->buffer[1];
-		if (buffer != nullptr) {
+		if (buffer != null) {
 			buffer->Release();
 		}
 	}
@@ -808,11 +807,11 @@ enum audio::orchestra::error audio::orchestra::api::Ds::closeStream() {
 	}
 	// Stop the callback thread.
 	m_private->threadRunning = false;
-	if (m_private->thread != nullptr) {
+	if (m_private->thread != null) {
 		m_private->thread->join();
-		m_private->thread = nullptr;
+		m_private->thread = null;
 	}
-	if (m_private->buffer[0]) { // the object pointer can be nullptr and valid
+	if (m_private->buffer[0]) { // the object pointer can be null and valid
 		LPDIRECTSOUND object = (LPDIRECTSOUND) m_private->id[0];
 		LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) m_private->buffer[0];
 		if (buffer) {
@@ -919,7 +918,7 @@ enum audio::orchestra::error audio::orchestra::api::Ds::stopStream() {
 		}
 		// Lock the buffer and clear it so that if we start to play again,
 		// we won't have old data playing.
-		result = buffer->Lock(0, m_private->dsBufferSize[0], &audioPtr, &dataLen, nullptr, nullptr, 0);
+		result = buffer->Lock(0, m_private->dsBufferSize[0], &audioPtr, &dataLen, null, null, 0);
 		if (FAILED(result)) {
 			ATA_ERROR(getErrorString(result) << ": locking output buffer!");
 			goto unlock;
@@ -927,7 +926,7 @@ enum audio::orchestra::error audio::orchestra::api::Ds::stopStream() {
 		// Zero the DS buffer
 		ZeroMemory(audioPtr, dataLen);
 		// Unlock the DS buffer
-		result = buffer->Unlock(audioPtr, dataLen, nullptr, 0);
+		result = buffer->Unlock(audioPtr, dataLen, null, 0);
 		if (FAILED(result)) {
 			ATA_ERROR(getErrorString(result) << ": unlocking output buffer!");
 			goto unlock;
@@ -938,7 +937,7 @@ enum audio::orchestra::error audio::orchestra::api::Ds::stopStream() {
 	if (    m_mode == audio::orchestra::mode_input
 	     || m_mode == audio::orchestra::mode_duplex) {
 		LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) m_private->buffer[1];
-		audioPtr = nullptr;
+		audioPtr = null;
 		dataLen = 0;
 		m_state = audio::orchestra::state::stopped;
 		result = buffer->Stop();
@@ -948,7 +947,7 @@ enum audio::orchestra::error audio::orchestra::api::Ds::stopStream() {
 		}
 		// Lock the buffer and clear it so that if we start to play again,
 		// we won't have old data playing.
-		result = buffer->Lock(0, m_private->dsBufferSize[1], &audioPtr, &dataLen, nullptr, nullptr, 0);
+		result = buffer->Lock(0, m_private->dsBufferSize[1], &audioPtr, &dataLen, null, null, 0);
 		if (FAILED(result)) {
 			ATA_ERROR(getErrorString(result) << ": locking input buffer!");
 			goto unlock;
@@ -956,7 +955,7 @@ enum audio::orchestra::error audio::orchestra::api::Ds::stopStream() {
 		// Zero the DS buffer
 		ZeroMemory(audioPtr, dataLen);
 		// Unlock the DS buffer
-		result = buffer->Unlock(audioPtr, dataLen, nullptr, 0);
+		result = buffer->Unlock(audioPtr, dataLen, null, 0);
 		if (FAILED(result)) {
 			ATA_ERROR(getErrorString(result) << ": unlocking input buffer!");
 			goto unlock;
@@ -986,7 +985,8 @@ enum audio::orchestra::error audio::orchestra::api::Ds::abortStream() {
 
 void audio::orchestra::api::Ds::callbackEvent() {
 	ethread::setName("DS IO-" + m_name);
-	if (m_state == audio::orchestra::state::stopped || m_state == audio::orchestra::state_stopping) {
+	if (    m_state == audio::orchestra::state::stopped
+	     || m_state == audio::orchestra::state::stopping) {
 		Sleep(50); // sleep 50 milliseconds
 		return;
 	}
@@ -1008,15 +1008,15 @@ void audio::orchestra::api::Ds::callbackEvent() {
 	// draining stream.
 	if (m_private->drainCounter == 0) {
 		audio::Time streamTime = getStreamTime();
-		std::vector<audio::orchestra::status> status;
+		etk::Vector<audio::orchestra::status> status;
 		if (    m_mode != audio::orchestra::mode_input
 		     && m_private->xrun[0] == true) {
-			status.push_back(audio::orchestra::status::underflow);
+			status.pushBack(audio::orchestra::status::underflow);
 			m_private->xrun[0] = false;
 		}
 		if (    m_mode != audio::orchestra::mode_output
 		     && m_private->xrun[1] == true) {
-			status.push_back(audio::orchestra::status::overflow);
+			status.pushBack(audio::orchestra::status::overflow);
 			m_private->xrun[1] = false;
 		}
 		int32_t cbReturnValue = m_callback(&m_userBuffer[1][0],
@@ -1039,8 +1039,8 @@ void audio::orchestra::api::Ds::callbackEvent() {
 	DWORD currentWritePointer, safeWritePointer;
 	DWORD currentReadPointer, safeReadPointer;
 	UINT nextWritePointer;
-	LPVOID buffer1 = nullptr;
-	LPVOID buffer2 = nullptr;
+	LPVOID buffer1 = null;
+	LPVOID buffer2 = null;
 	DWORD bufferSize1 = 0;
 	DWORD bufferSize2 = 0;
 	char *buffer;
@@ -1063,23 +1063,23 @@ void audio::orchestra::api::Ds::callbackEvent() {
 			LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) m_private->buffer[0];
 			LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) m_private->buffer[1];
 			DWORD startSafeWritePointer, startSafeReadPointer;
-			result = dsWriteBuffer->GetCurrentPosition(nullptr, &startSafeWritePointer);
+			result = dsWriteBuffer->GetCurrentPosition(null, &startSafeWritePointer);
 			if (FAILED(result)) {
 				ATA_ERROR(getErrorString(result) << ": getting current write position!");
 				return;
 			}
-			result = dsCaptureBuffer->GetCurrentPosition(nullptr, &startSafeReadPointer);
+			result = dsCaptureBuffer->GetCurrentPosition(null, &startSafeReadPointer);
 			if (FAILED(result)) {
 				ATA_ERROR(getErrorString(result) << ": getting current read position!");
 				return;
 			}
 			while (true) {
-				result = dsWriteBuffer->GetCurrentPosition(nullptr, &safeWritePointer);
+				result = dsWriteBuffer->GetCurrentPosition(null, &safeWritePointer);
 				if (FAILED(result)) {
 					ATA_ERROR(getErrorString(result) << ": getting current write position!");
 					return;
 				}
-				result = dsCaptureBuffer->GetCurrentPosition(nullptr, &safeReadPointer);
+				result = dsCaptureBuffer->GetCurrentPosition(null, &safeReadPointer);
 				if (FAILED(result)) {
 					ATA_ERROR(getErrorString(result) << ": getting current read position!");
 					return;
@@ -1153,7 +1153,7 @@ void audio::orchestra::api::Ds::callbackEvent() {
 			// safeWritePointer and leadPointer.	If leadPointer is not
 			// beyond the next endWrite position, wait until it is.
 			leadPointer = safeWritePointer + m_private->dsPointerLeadTime[0];
-			//std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl;
+			//ATA_DEBUG("safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer);
 			if (leadPointer > dsBufferSize) {
 				leadPointer -= dsBufferSize;
 			}
@@ -1200,7 +1200,7 @@ void audio::orchestra::api::Ds::callbackEvent() {
 		}
 		// Copy our buffer into the DS buffer
 		CopyMemory(buffer1, buffer, bufferSize1);
-		if (buffer2 != nullptr) {
+		if (buffer2 != null) {
 			CopyMemory(buffer2, buffer+bufferSize1, bufferSize2);
 		}
 		// Update our buffer offset and unlock sound buffer
@@ -1318,12 +1318,12 @@ void audio::orchestra::api::Ds::callbackEvent() {
 		if (m_duplexPrerollBytes <= 0) {
 			// Copy our buffer into the DS buffer
 			CopyMemory(buffer, buffer1, bufferSize1);
-			if (buffer2 != nullptr) {
+			if (buffer2 != null) {
 				CopyMemory(buffer+bufferSize1, buffer2, bufferSize2);
 			}
 		} else {
 			memset(buffer, 0, bufferSize1);
-			if (buffer2 != nullptr) {
+			if (buffer2 != null) {
 				memset(buffer + bufferSize1, 0, bufferSize2);
 			}
 			m_duplexPrerollBytes -= bufferSize1 + bufferSize2;

audio/orchestra/api/Ds.hpp

@@ -17,7 +17,7 @@ namespace audio {
 				public:
 					Ds();
 					virtual ~Ds();
-					const std::string& getCurrentApi() {
+					const etk::String& getCurrentApi() {
 						return audio::orchestra::typeDs;
 					}
 					uint32_t getDeviceCount();

audio/orchestra/api/Dummy.cpp

@@ -6,11 +6,11 @@
  */
 
 #if defined(ORCHESTRA_BUILD_DUMMY)
-#include <audio/orchestra/api/Dummy.h>
-#include <audio/orchestra/debug.h>
+#include <audio/orchestra/api/Dummy.hpp>
+#include <audio/orchestra/debug.hpp>
 
 ememory::SharedPtr<audio::orchestra::Api> audio::orchestra::api::Dummy::create() {
-	return ememory::SharedPtr<audio::orchestra::api::Dummy>(new audio::orchestra::api::Dummy());
+	return ememory::SharedPtr<audio::orchestra::api::Dummy>(ETK_NEW(audio::orchestra::api::Dummy));
 }
 
 

audio/orchestra/api/Dummy.hpp

@@ -8,7 +8,7 @@
 
 #ifdef ORCHESTRA_BUILD_DUMMY
 
-#include <audio/orchestra/Interface.h>
+#include <audio/orchestra/Interface.hpp>
 
 namespace audio {
 	namespace orchestra {
@@ -18,7 +18,7 @@ namespace audio {
 					static ememory::SharedPtr<audio::orchestra::Api> create();
 				public:
 					Dummy();
-					const std::string& getCurrentApi() {
+					const etk::String& getCurrentApi() {
 						return audio::orchestra::typeDummy;
 					}
 					uint32_t getDeviceCount();

audio/orchestra/api/Jack.cpp

@@ -7,17 +7,17 @@
 
 // must run before :          
 #if defined(ORCHESTRA_BUILD_JACK)
-#include <unistd.h>
-#include <limits.h>
-#include <iostream>
-#include <audio/orchestra/Interface.h>
-#include <audio/orchestra/debug.h>
-#include <string.h>
-#include <ethread/tools.h>
-#include <audio/orchestra/api/Jack.h>
+extern "C" {
+	#include <limits.h>
+	#include <string.h>
+}
+#include <audio/orchestra/Interface.hpp>
+#include <audio/orchestra/debug.hpp>
+#include <ethread/tools.hpp>
+#include <audio/orchestra/api/Jack.hpp>
 
 ememory::SharedPtr<audio::orchestra::Api> audio::orchestra::api::Jack::create() {
-	return ememory::SharedPtr<audio::orchestra::api::Jack>(new audio::orchestra::api::Jack());
+	return ememory::SharedPtr<audio::orchestra::api::Jack>(ETK_NEW(audio::orchestra::api::Jack));
 }
 
 
@@ -52,9 +52,9 @@ ememory::SharedPtr<audio::orchestra::Api> audio::orchestra::api::Jack::create()
 // stream cannot be opened.
 
 #include <jack/jack.h>
-#include <unistd.h>
-#include <cstdio>
-
+extern "C" {
+#include <stdio.h>
+}
 
 namespace audio {
 	namespace orchestra {
@@ -63,9 +63,9 @@ namespace audio {
 				public:
 					jack_client_t *client;
 					jack_port_t **ports[2];
-					std::string deviceName[2];
+					etk::String deviceName[2];
 					bool xrun[2];
-					std::condition_variable condition;
+					ethread::Semaphore m_semaphore;
 					int32_t drainCounter; // Tracks callback counts when draining
 					bool internalDrain; // Indicates if stop is initiated from callback or not.
 					
@@ -84,7 +84,7 @@ namespace audio {
 }
 
 audio::orchestra::api::Jack::Jack() :
-  m_private(new audio::orchestra::api::JackPrivate()) {
+  m_private(ETK_NEW(audio::orchestra::api::JackPrivate)) {
 	// Nothing to do here.
 }
 
@@ -97,23 +97,23 @@ audio::orchestra::api::Jack::~Jack() {
 uint32_t audio::orchestra::api::Jack::getDeviceCount() {
 	// See if we can become a jack client.
 	jack_options_t options = (jack_options_t) (JackNoStartServer); //JackNullOption;
-	jack_status_t *status = nullptr;
+	jack_status_t *status = null;
 	jack_client_t *client = jack_client_open("orchestraJackCount", options, status);
-	if (client == nullptr) {
+	if (client == null) {
 		return 0;
 	}
 	const char **ports;
-	std::string port, previousPort;
+	etk::String port, previousPort;
 	uint32_t nChannels = 0, nDevices = 0;
-	ports = jack_get_ports(client, nullptr, nullptr, 0);
+	ports = jack_get_ports(client, null, null, 0);
 	if (ports) {
 		// Parse the port names up to the first colon (:).
 		size_t iColon = 0;
 		do {
 			port = (char *) ports[ nChannels ];
 			iColon = port.find(":");
-			if (iColon != std::string::npos) {
-				port = port.substr(0, iColon + 1);
+			if (iColon != etk::String::npos) {
+				port = port.extract(0, iColon + 1);
 				if (port != previousPort) {
 					nDevices++;
 					previousPort = port;
@@ -121,6 +121,7 @@ uint32_t audio::orchestra::api::Jack::getDeviceCount() {
 			}
 		} while (ports[++nChannels]);
 		free(ports);
+		ports = null;
 	}
 	jack_client_close(client);
 	return nDevices*2;
@@ -129,18 +130,18 @@ uint32_t audio::orchestra::api::Jack::getDeviceCount() {
 audio::orchestra::DeviceInfo audio::orchestra::api::Jack::getDeviceInfo(uint32_t _device) {
 	audio::orchestra::DeviceInfo info;
 	jack_options_t options = (jack_options_t) (JackNoStartServer); //JackNullOption
-	jack_status_t *status = nullptr;
+	jack_status_t *status = null;
 	jack_client_t *client = jack_client_open("orchestraJackInfo", options, status);
-	if (client == nullptr) {
+	if (client == null) {
 		ATA_ERROR("Jack server not found or connection error!");
 		// TODO : audio::orchestra::error_warning;
 		info.clear();
 		return info;
 	}
 	const char **ports;
-	std::string port, previousPort;
+	etk::String port, previousPort;
 	uint32_t nPorts = 0, nDevices = 0;
-	ports = jack_get_ports(client, nullptr, nullptr, 0);
+	ports = jack_get_ports(client, null, null, 0);
 	int32_t deviceID = _device/2;
 	info.input = _device%2==0?true:false; // note that jack sens are inverted
 	if (ports) {
@@ -149,8 +150,8 @@ audio::orchestra::DeviceInfo audio::orchestra::api::Jack::getDeviceInfo(uint32_t
 		do {
 			port = (char *) ports[nPorts];
 			iColon = port.find(":");
-			if (iColon != std::string::npos) {
-				port = port.substr(0, iColon);
+			if (iColon != etk::String::npos) {
+				port = port.extract(0, iColon);
 				if (port != previousPort) {
 					if (nDevices == deviceID) {
 						info.name = port;
@@ -170,25 +171,25 @@ audio::orchestra::DeviceInfo audio::orchestra::api::Jack::getDeviceInfo(uint32_t
 	}
 	// Get the current jack server sample rate.
 	info.sampleRates.clear();
-	info.sampleRates.push_back(jack_get_sample_rate(client));
+	info.sampleRates.pushBack(jack_get_sample_rate(client));
 	if (info.input == true) {
-		ports = jack_get_ports(client, info.name.c_str(), nullptr, JackPortIsOutput);
+		ports = jack_get_ports(client, info.name.c_str(), null, JackPortIsOutput);
 		if (ports) {
 			int32_t iii=0;
 			while (ports[iii]) {
 				ATA_ERROR(" ploppp='" << ports[iii] << "'");
-				info.channels.push_back(audio::channel_unknow);
+				info.channels.pushBack(audio::channel_unknow);
 				iii++;
 			}
 			free(ports);
 		}
 	} else {
-		ports = jack_get_ports(client, info.name.c_str(), nullptr, JackPortIsInput);
+		ports = jack_get_ports(client, info.name.c_str(), null, JackPortIsInput);
 		if (ports) {
 			int32_t iii=0;
 			while (ports[iii]) {
 				ATA_ERROR(" ploppp='" << ports[iii] << "'");
-				info.channels.push_back(audio::channel_unknow);
+				info.channels.pushBack(audio::channel_unknow);
 				iii++;
 			}
 			free(ports);
@@ -202,7 +203,7 @@ audio::orchestra::DeviceInfo audio::orchestra::api::Jack::getDeviceInfo(uint32_t
 		return info;
 	}
 	// Jack always uses 32-bit floats.
-	info.nativeFormats.push_back(audio::format_float);
+	info.nativeFormats.pushBack(audio::format_float);
 	// Jack doesn't provide default devices so we'll use the first available one.
 	if (deviceID == 0) {
 		info.isDefault = true;
@@ -223,15 +224,6 @@ int32_t audio::orchestra::api::Jack::jackCallbackHandler(jack_nframes_t _nframes
 	return 0;
 }
 
-// This function will be called by a spawned thread when the Jack
-// server signals that it is shutting down.	It is necessary to handle
-// it this way because the jackShutdown() function must return before
-// the jack_deactivate() function (in closeStream()) will return.
-void audio::orchestra::api::Jack::jackCloseStream(void* _userData) {
-	ethread::setName("Jack_closeStream");
-	audio::orchestra::api::Jack* myClass = reinterpret_cast<audio::orchestra::api::Jack*>(_userData);
-	myClass->closeStream();
-}
 
 void audio::orchestra::api::Jack::jackShutdown(void* _userData) {
 	audio::orchestra::api::Jack* myClass = reinterpret_cast<audio::orchestra::api::Jack*>(_userData);
@@ -243,7 +235,7 @@ void audio::orchestra::api::Jack::jackShutdown(void* _userData) {
 	if (myClass->isStreamRunning() == false) {
 		return;
 	}
-	new std::thread(&audio::orchestra::api::Jack::jackCloseStream, _userData);
+	ETK_NEW(ethread::Thread, [=](){myClass->closeStream();});
 	ATA_ERROR("The Jack server is shutting down this client ... stream stopped and closed!!");
 }
 
@@ -272,8 +264,8 @@ bool audio::orchestra::api::Jack::open(uint32_t _device,
 	     || (    _mode == audio::orchestra::mode_input
 	          && m_mode != audio::orchestra::mode_output)) {
 		jack_options_t jackoptions = (jack_options_t) (JackNoStartServer); //JackNullOption;
-		jack_status_t *status = nullptr;
-		if (!_options.streamName.empty()) {
+		jack_status_t *status = null;
+		if (_options.streamName.size() != 0) {
 			client = jack_client_open(_options.streamName.c_str(), jackoptions, status);
 		} else {
 			client = jack_client_open("orchestraJack", jackoptions, status);
@@ -287,19 +279,19 @@ bool audio::orchestra::api::Jack::open(uint32_t _device,
 		client = m_private->client;
 	}
 	const char **ports;
-	std::string port, previousPort, deviceName;
+	etk::String port, previousPort, deviceName;
 	uint32_t nPorts = 0, nDevices = 0;
 	int32_t deviceID = _device/2;
 	bool isInput = _device%2==0?true:false;
-	ports = jack_get_ports(client, nullptr, nullptr, 0);
+	ports = jack_get_ports(client, null, null, 0);
 	if (ports) {
 		// Parse the port names up to the first colon (:).
 		size_t iColon = 0;
 		do {
 			port = (char *) ports[ nPorts ];
 			iColon = port.find(":");
-			if (iColon != std::string::npos) {
-				port = port.substr(0, iColon);
+			if (iColon != etk::String::npos) {
+				port = port.extract(0, iColon);
 				if (port != previousPort) {
 					if (nDevices == deviceID) {
 						deviceName = port;
@@ -322,7 +314,7 @@ bool audio::orchestra::api::Jack::open(uint32_t _device,
 	if (_mode == audio::orchestra::mode_input) {
 		flag = JackPortIsOutput;
 	}
-	ports = jack_get_ports(client, deviceName.c_str(), nullptr, flag);
+	ports = jack_get_ports(client, deviceName.c_str(), null, flag);
 	if (ports) {
 		while (ports[ nChannels ]) {
 			nChannels++;
@@ -343,7 +335,7 @@ bool audio::orchestra::api::Jack::open(uint32_t _device,
 	}
 	m_sampleRate = jackRate;
 	// Get the latency of the JACK port.
-	ports = jack_get_ports(client, deviceName.c_str(), nullptr, flag);
+	ports = jack_get_ports(client, deviceName.c_str(), null, flag);
 	if (ports[ _firstChannel ]) {
 		// Added by Ge Wang
 		jack_latency_callback_mode_t cbmode = (_mode == audio::orchestra::mode_input ? JackCaptureLatency : JackPlaybackLatency);
@@ -409,7 +401,7 @@ bool audio::orchestra::api::Jack::open(uint32_t _device,
 			bufferBytes *= *_bufferSize;
 			if (m_deviceBuffer) free(m_deviceBuffer);
 			m_deviceBuffer = (char *) calloc(bufferBytes, 1);
-			if (m_deviceBuffer == nullptr) {
+			if (m_deviceBuffer == null) {
 				ATA_ERROR("error allocating device buffer memory.");
 				goto error;
 			}
@@ -417,7 +409,7 @@ bool audio::orchestra::api::Jack::open(uint32_t _device,
 	}
 	// Allocate memory for the Jack ports (channels) identifiers.
 	m_private->ports[modeToIdTable(_mode)] = (jack_port_t **) malloc (sizeof (jack_port_t *) * _channels);
-	if (m_private->ports[modeToIdTable(_mode)] == nullptr)	{
+	if (m_private->ports[modeToIdTable(_mode)] == null)	{
 		ATA_ERROR("error allocating port memory.");
 		goto error;
 	}
@@ -464,20 +456,20 @@ bool audio::orchestra::api::Jack::open(uint32_t _device,
 	return true;
 error:
 	jack_client_close(m_private->client);
-	if (m_private->ports[0] != nullptr) {
+	if (m_private->ports[0] != null) {
 		free(m_private->ports[0]);
-		m_private->ports[0] = nullptr;
+		m_private->ports[0] = null;
 	}
-	if (m_private->ports[1] != nullptr) {
+	if (m_private->ports[1] != null) {
 		free(m_private->ports[1]);
-		m_private->ports[1] = nullptr;
+		m_private->ports[1] = null;
 	}
 	for (int32_t iii=0; iii<2; ++iii) {
 		m_userBuffer[iii].clear();
 	}
 	if (m_deviceBuffer) {
 		free(m_deviceBuffer);
-		m_deviceBuffer = nullptr;
+		m_deviceBuffer = null;
 	}
 	return false;
 }
@@ -487,26 +479,26 @@ enum audio::orchestra::error audio::orchestra::api::Jack::closeStream() {
 		ATA_ERROR("no open stream to close!");
 		return audio::orchestra::error_warning;
 	}
-	if (m_private != nullptr) {
+	if (m_private != null) {
 		if (m_state == audio::orchestra::state::running) {
 			jack_deactivate(m_private->client);
 		}
 		jack_client_close(m_private->client);
 	}
-	if (m_private->ports[0] != nullptr) {
+	if (m_private->ports[0] != null) {
 		free(m_private->ports[0]);
-		m_private->ports[0] = nullptr;
+		m_private->ports[0] = null;
 	}
-	if (m_private->ports[1] != nullptr) {
+	if (m_private->ports[1] != null) {
 		free(m_private->ports[1]);
-		m_private->ports[1] = nullptr;
+		m_private->ports[1] = null;
 	}
 	for (int32_t i=0; i<2; i++) {
 		m_userBuffer[i].clear();
 	}
 	if (m_deviceBuffer) {
 		free(m_deviceBuffer);
-		m_deviceBuffer = nullptr;
+		m_deviceBuffer = null;
 	}
 	m_mode = audio::orchestra::mode_unknow;
 	m_state = audio::orchestra::state::closed;
@@ -533,8 +525,8 @@ enum audio::orchestra::error audio::orchestra::api::Jack::startStream() {
 	if (    m_mode == audio::orchestra::mode_output
 	     || m_mode == audio::orchestra::mode_duplex) {
 		result = 1;
-		ports = jack_get_ports(m_private->client, m_private->deviceName[0].c_str(), nullptr, JackPortIsInput);
-		if (ports == nullptr) {
+		ports = jack_get_ports(m_private->client, m_private->deviceName[0].c_str(), null, JackPortIsInput);
+		if (ports == null) {
 			ATA_ERROR("error determining available JACK input ports!");
 			goto unlock;
 		}
@@ -556,8 +548,8 @@ enum audio::orchestra::error audio::orchestra::api::Jack::startStream() {
 	if (    m_mode == audio::orchestra::mode_input
 	     || m_mode == audio::orchestra::mode_duplex) {
 		result = 1;
-		ports = jack_get_ports(m_private->client, m_private->deviceName[1].c_str(), nullptr, JackPortIsOutput);
-		if (ports == nullptr) {
+		ports = jack_get_ports(m_private->client, m_private->deviceName[1].c_str(), null, JackPortIsOutput);
+		if (ports == null) {
 			ATA_ERROR("error determining available JACK output ports!");
 			goto unlock;
 		}
@@ -597,8 +589,7 @@ enum audio::orchestra::error audio::orchestra::api::Jack::stopStream() {
 	     || m_mode == audio::orchestra::mode_duplex) {
 		if (m_private->drainCounter == 0) {
 			m_private->drainCounter = 2;
-			std::unique_lock<std::mutex> lck(m_mutex);
-			m_private->condition.wait(lck);
+			m_private->m_semaphore.wait();
 		}
 	}
 	jack_deactivate(m_private->client);
@@ -618,17 +609,6 @@ enum audio::orchestra::error audio::orchestra::api::Jack::abortStream() {
 	return stopStream();
 }
 
-// This function will be called by a spawned thread when the user
-// callback function signals that the stream should be stopped or
-// aborted.	It is necessary to handle it this way because the
-// callbackEvent() function must return before the jack_deactivate()
-// function will return.
-static void jackStopStream(void* _userData) {
-	ethread::setName("Jack_stopStream");
-	audio::orchestra::api::Jack* myClass = reinterpret_cast<audio::orchestra::api::Jack*>(_userData);
-	myClass->stopStream();
-}
-
 bool audio::orchestra::api::Jack::callbackEvent(uint64_t _nframes) {
 	if (    m_state == audio::orchestra::state::stopped
 	     || m_state == audio::orchestra::state::stopping) {
@@ -646,22 +626,22 @@ bool audio::orchestra::api::Jack::callbackEvent(uint64_t _nframes) {
 	if (m_private->drainCounter > 3) {
 		m_state = audio::orchestra::state::stopping;
 		if (m_private->internalDrain == true) {
-			new std::thread(jackStopStream, this);
+			ETK_NEW(ethread::Thread, [&](){stopStream();}, "Jack_stopStream");
 		} else {
-			m_private->condition.notify_one();
+			m_private->m_semaphore.post();
 		}
 		return true;
 	}
 	// Invoke user callback first, to get fresh output data.
 	if (m_private->drainCounter == 0) {
 		audio::Time streamTime = getStreamTime();
-		std::vector<enum audio::orchestra::status> status;
+		etk::Vector<enum audio::orchestra::status> status;
 		if (m_mode != audio::orchestra::mode_input && m_private->xrun[0] == true) {
-			status.push_back(audio::orchestra::status::underflow);
+			status.pushBack(audio::orchestra::status::underflow);
 			m_private->xrun[0] = false;
 		}
 		if (m_mode != audio::orchestra::mode_output && m_private->xrun[1] == true) {
-			status.push_back(audio::orchestra::status::overflow);
+			status.pushBack(audio::orchestra::status::overflow);
 			m_private->xrun[1] = false;
 		}
 		int32_t cbReturnValue = m_callback(&m_userBuffer[1][0],
@@ -673,7 +653,7 @@ bool audio::orchestra::api::Jack::callbackEvent(uint64_t _nframes) {
 		if (cbReturnValue == 2) {
 			m_state = audio::orchestra::state::stopping;
 			m_private->drainCounter = 2;
-			new std::thread(jackStopStream, this);
+			ETK_NEW(ethread::Thread, [&](){stopStream();}, "Jack_stopStream2");
 			return true;
 		}
 		else if (cbReturnValue == 1) {

audio/orchestra/api/Jack.hpp

@@ -19,7 +19,7 @@ namespace audio {
 				public:
 					Jack();
 					virtual ~Jack();
-					const std::string& getCurrentApi() {
+					const etk::String& getCurrentApi() {
 						return audio::orchestra::typeJack;
 					}
 					uint32_t getDeviceCount();
@@ -36,7 +36,6 @@ namespace audio {
 					bool callbackEvent(uint64_t _nframes);
 				private:
 					static int32_t jackXrun(void* _userData);
-					static void jackCloseStream(void* _userData);
 					static void jackShutdown(void* _userData);
 					static int32_t jackCallbackHandler(jack_nframes_t _nframes, void* _userData);
 				private:

audio/orchestra/api/Pulse.cpp

@@ -8,19 +8,20 @@
 
 #if defined(ORCHESTRA_BUILD_PULSE)
 
-#include <unistd.h>
-#include <limits.h>
-#include <audio/orchestra/Interface.h>
-#include <audio/orchestra/debug.h>
+extern "C" {
+	#include <limits.h>
+	#include <stdio.h>
+}
+#include <audio/orchestra/Interface.hpp>
+#include <audio/orchestra/debug.hpp>
 #include <pulse/error.h>
 #include <pulse/simple.h>
-#include <cstdio>
-#include <ethread/tools.h>
-#include <audio/orchestra/api/PulseDeviceList.h>
-#include <audio/orchestra/api/Pulse.h>
+#include <ethread/tools.hpp>
+#include <audio/orchestra/api/PulseDeviceList.hpp>
+#include <audio/orchestra/api/Pulse.hpp>
 
 ememory::SharedPtr<audio::orchestra::Api> audio::orchestra::api::Pulse::create() {
-	return ememory::SharedPtr<audio::orchestra::api::Pulse>(new audio::orchestra::api::Pulse());
+	return ememory::SharedPtr<audio::orchestra::api::Pulse>(ETK_NEW(audio::orchestra::api::Pulse));
 }
 
 
@@ -53,9 +54,9 @@ namespace audio {
 			class PulsePrivate {
 				public:
 					pa_simple* handle;
-					ememory::SharedPtr<std::thread> thread;
+					ememory::SharedPtr<ethread::Thread> thread;
 					bool threadRunning;
-					std::condition_variable runnable_cv;
+					ethread::Semaphore m_semaphore;
 					bool runnable;
 					PulsePrivate() :
 					  handle(0),
@@ -68,7 +69,7 @@ namespace audio {
 	}
 }
 audio::orchestra::api::Pulse::Pulse() :
-  m_private(new audio::orchestra::api::PulsePrivate()) {
+  m_private(ETK_NEW(audio::orchestra::api::PulsePrivate)) {
 	
 }
 
@@ -80,7 +81,7 @@ audio::orchestra::api::Pulse::~Pulse() {
 
 uint32_t audio::orchestra::api::Pulse::getDeviceCount() {
 	#if 1
-		std::vector<audio::orchestra::DeviceInfo> list = audio::orchestra::api::pulse::getDeviceList();
+		etk::Vector<audio::orchestra::DeviceInfo> list = audio::orchestra::api::pulse::getDeviceList();
 		return list.size();
 	#else
 		return 1;
@@ -88,7 +89,7 @@ uint32_t audio::orchestra::api::Pulse::getDeviceCount() {
 }
 
 audio::orchestra::DeviceInfo audio::orchestra::api::Pulse::getDeviceInfo(uint32_t _device) {
-	std::vector<audio::orchestra::DeviceInfo> list = audio::orchestra::api::pulse::getDeviceList();
+	etk::Vector<audio::orchestra::DeviceInfo> list = audio::orchestra::api::pulse::getDeviceList();
 	if (_device >= list.size()) {
 		ATA_ERROR("Request device out of IDs:" << _device << " >= " << list.size());
 		return audio::orchestra::DeviceInfo();
@@ -96,10 +97,7 @@ audio::orchestra::DeviceInfo audio::orchestra::api::Pulse::getDeviceInfo(uint32_
 	return list[_device];
 }
 
-static void pulseaudio_callback(void* _userData) {
-	audio::orchestra::api::Pulse* myClass = reinterpret_cast<audio::orchestra::api::Pulse*>(_userData);
-	myClass->callbackEvent();
-}
+
 
 void audio::orchestra::api::Pulse::callbackEvent() {
 	ethread::setName("Pulse IO-" + m_name);
@@ -113,15 +111,15 @@ enum audio::orchestra::error audio::orchestra::api::Pulse::closeStream() {
 	m_mutex.lock();
 	if (m_state == audio::orchestra::state::stopped) {
 		m_private->runnable = true;
-		m_private->runnable_cv.notify_one();;
+		m_private->m_semaphore.post();;
 	}
-	m_mutex.unlock();
+	m_mutex.unLock();
 	m_private->thread->join();
 	if (m_mode == audio::orchestra::mode_output) {
-		pa_simple_flush(m_private->handle, nullptr);
+		pa_simple_flush(m_private->handle, null);
 	}
 	pa_simple_free(m_private->handle);
-	m_private->handle = nullptr;
+	m_private->handle = null;
 	m_userBuffer[0].clear();
 	m_userBuffer[1].clear();
 	m_state = audio::orchestra::state::closed;
@@ -131,12 +129,11 @@ enum audio::orchestra::error audio::orchestra::api::Pulse::closeStream() {
 
 void audio::orchestra::api::Pulse::callbackEventOneCycle() {
 	if (m_state == audio::orchestra::state::stopped) {
-		std::unique_lock<std::mutex> lck(m_mutex);
 		while (!m_private->runnable) {
-			m_private->runnable_cv.wait(lck);
+			m_private->m_semaphore.wait();
 		}
 		if (m_state != audio::orchestra::state::running) {
-			m_mutex.unlock();
+			m_mutex.unLock();
 			return;
 		}
 	}
@@ -145,7 +142,7 @@ void audio::orchestra::api::Pulse::callbackEventOneCycle() {
 		return;
 	}
 	audio::Time streamTime = getStreamTime();
-	std::vector<enum audio::orchestra::status> status;
+	etk::Vector<enum audio::orchestra::status> status;
 	int32_t doStopStream = m_callback(&m_userBuffer[audio::orchestra::modeToIdTable(audio::orchestra::mode_input)][0],
 	                                  streamTime,
 	                                  &m_userBuffer[audio::orchestra::modeToIdTable(audio::orchestra::mode_output)][0],
@@ -160,7 +157,7 @@ void audio::orchestra::api::Pulse::callbackEventOneCycle() {
 	void *pulse_in = m_doConvertBuffer[audio::orchestra::modeToIdTable(audio::orchestra::mode_input)] ? m_deviceBuffer : &m_userBuffer[audio::orchestra::modeToIdTable(audio::orchestra::mode_input)][0];
 	void *pulse_out = m_doConvertBuffer[audio::orchestra::modeToIdTable(audio::orchestra::mode_output)] ? m_deviceBuffer : &m_userBuffer[audio::orchestra::modeToIdTable(audio::orchestra::mode_output)][0];
 	if (m_state != audio::orchestra::state::running) {
-		goto unlock;
+		goto unLock;
 	}
 	int32_t pa_error;
 	size_t bytes;
@@ -194,8 +191,8 @@ void audio::orchestra::api::Pulse::callbackEventOneCycle() {
 			              m_convertInfo[audio::orchestra::modeToIdTable(audio::orchestra::mode_input)]);
 		}
 	}
-unlock:
-	m_mutex.unlock();
+unLock:
+	m_mutex.unLock();
 	audio::orchestra::Api::tickStreamTime();
 	if (doStopStream == 1) {
 		stopStream();
@@ -218,8 +215,8 @@ enum audio::orchestra::error audio::orchestra::api::Pulse::startStream() {
 	m_mutex.lock();
 	m_state = audio::orchestra::state::running;
 	m_private->runnable = true;
-	m_private->runnable_cv.notify_one();
-	m_mutex.unlock();
+	m_private->m_semaphore.post();
+	m_mutex.unLock();
 	return audio::orchestra::error_none;
 }
 
@@ -234,18 +231,18 @@ enum audio::orchestra::error audio::orchestra::api::Pulse::stopStream() {
 	}
 	m_state = audio::orchestra::state::stopped;
 	m_mutex.lock();
-	if (    m_private != nullptr
-	     && m_private->handle != nullptr
+	if (    m_private != null
+	     && m_private->handle != null
 	     && m_mode == audio::orchestra::mode_output) {
 		int32_t pa_error;
 		if (pa_simple_drain(m_private->handle, &pa_error) < 0) {
 			ATA_ERROR("error draining output device, " << pa_strerror(pa_error) << ".");
-			m_mutex.unlock();
+			m_mutex.unLock();
 			return audio::orchestra::error_systemError;
 		}
 	}
 	m_state = audio::orchestra::state::stopped;
-	m_mutex.unlock();
+	m_mutex.unLock();
 	return audio::orchestra::error_none;
 }
 
@@ -260,18 +257,18 @@ enum audio::orchestra::error audio::orchestra::api::Pulse::abortStream() {
 	}
 	m_state = audio::orchestra::state::stopped;
 	m_mutex.lock();
-	if (    m_private != nullptr
-	     && m_private->handle != nullptr
+	if (    m_private != null
+	     && m_private->handle != null
 	     && m_mode == audio::orchestra::mode_output) {
 		int32_t pa_error;
 		if (pa_simple_flush(m_private->handle, &pa_error) < 0) {
 			ATA_ERROR("error flushing output device, " << pa_strerror(pa_error) << ".");
-			m_mutex.unlock();
+			m_mutex.unLock();
 			return audio::orchestra::error_systemError;
 		}
 	}
 	m_state = audio::orchestra::state::stopped;
-	m_mutex.unlock();
+	m_mutex.unLock();
 	return audio::orchestra::error_none;
 }
 
@@ -356,7 +353,7 @@ bool audio::orchestra::api::Pulse::open(uint32_t _device,
 			bufferBytes *= *_bufferSize;
 			if (m_deviceBuffer) free(m_deviceBuffer);
 			m_deviceBuffer = (char *) calloc(bufferBytes, 1);
-			if (m_deviceBuffer == nullptr) {
+			if (m_deviceBuffer == null) {
 				ATA_ERROR("error allocating device buffer memory.");
 				goto error;
 			}
@@ -370,15 +367,15 @@ bool audio::orchestra::api::Pulse::open(uint32_t _device,
 	int32_t error;
 	switch (_mode) {
 		case audio::orchestra::mode_input:
-			m_private->handle = pa_simple_new(nullptr, "orchestra", PA_STREAM_RECORD, nullptr, "Record", &ss, nullptr, nullptr, &error);
-			if (m_private->handle == nullptr) {
+			m_private->handle = pa_simple_new(null, "orchestra", PA_STREAM_RECORD, null, "Record", &ss, null, null, &error);
+			if (m_private->handle == null) {
 				ATA_ERROR("error connecting input to PulseAudio server.");
 				goto error;
 			}
 			break;
 		case audio::orchestra::mode_output:
-			m_private->handle = pa_simple_new(nullptr, "orchestra", PA_STREAM_PLAYBACK, nullptr, "Playback", &ss, nullptr, nullptr, &error);
-			if (m_private->handle == nullptr) {
+			m_private->handle = pa_simple_new(null, "orchestra", PA_STREAM_PLAYBACK, null, "Playback", &ss, null, null, &error);
+			if (m_private->handle == null) {
 				ATA_ERROR("error connecting output to PulseAudio server.");
 				goto error;
 			}
@@ -393,8 +390,8 @@ bool audio::orchestra::api::Pulse::open(uint32_t _device,
 	}
 	if (m_private->threadRunning == false) {
 		m_private->threadRunning = true;
-		m_private->thread = ememory::makeShared<std::thread>(&pulseaudio_callback, this);
-		if (m_private->thread == nullptr) {
+		m_private->thread = ememory::makeShared<ethread::Thread>([&](){callbackEvent();}, "pulseCallback");
+		if (m_private->thread == null) {
 			ATA_ERROR("error creating thread.");
 			goto error;
 		}

audio/orchestra/api/Pulse.hpp

@@ -17,7 +17,7 @@ namespace audio {
 				public:
 					Pulse();
 					virtual ~Pulse();
-					const std::string& getCurrentApi() {
+					const etk::String& getCurrentApi() {
 						return audio::orchestra::typePulse;
 					}
 					uint32_t getDeviceCount();
@@ -34,7 +34,7 @@ namespace audio {
 					void callbackEvent();
 				private:
 					ememory::SharedPtr<PulsePrivate> m_private;
-					std::vector<audio::orchestra::DeviceInfo> m_devices;
+					etk::Vector<audio::orchestra::DeviceInfo> m_devices;
 					void saveDeviceInfo();
 					bool open(uint32_t _device,
 					          audio::orchestra::mode _mode,

audio/orchestra/api/PulseDeviceList.cpp

@@ -6,16 +6,17 @@
  */
 
 #if defined(ORCHESTRA_BUILD_PULSE)
-
-#include <stdio.h>
-#include <string.h>
+extern "C" {
+	#include <stdio.h>
+	#include <string.h>
+}
 #include <pulse/pulseaudio.h>
-#include <audio/orchestra/api/PulseDeviceList.h>
-#include <audio/orchestra/debug.h>
-#include <audio/Time.h>
-#include <audio/Duration.h>
-#include <audio/format.h>
-#include <etk/stdTools.h>
+#include <audio/orchestra/api/PulseDeviceList.hpp>
+#include <audio/orchestra/debug.hpp>
+#include <audio/Time.hpp>
+#include <audio/Duration.hpp>
+#include <audio/format.hpp>
+#include <etk/stdTools.hpp>
 
 // This callback gets called when our context changes state.  We really only
 // care about when it's ready or if it has failed
@@ -105,101 +106,101 @@ static audio::format getFormatFromPulseFormat(enum pa_sample_format _format) {
 	return audio::format_unknow;
 }
 
-static std::vector<audio::channel> getChannelOrderFromPulseChannel(const struct pa_channel_map& _map) {
-	std::vector<audio::channel> out;
+static etk::Vector<audio::channel> getChannelOrderFromPulseChannel(const struct pa_channel_map& _map) {
+	etk::Vector<audio::channel> out;
 	
 	for (int32_t iii=0; iii<_map.channels; ++iii) {
 		switch(_map.map[iii]) {
 			default:
 			case PA_CHANNEL_POSITION_MAX:
 			case PA_CHANNEL_POSITION_INVALID:
-				out.push_back(audio::channel_unknow);
+				out.pushBack(audio::channel_unknow);
 				break;
 			case PA_CHANNEL_POSITION_MONO:
 			case PA_CHANNEL_POSITION_FRONT_CENTER:
-				out.push_back(audio::channel_frontCenter);
+				out.pushBack(audio::channel_frontCenter);
 				break;
 			case PA_CHANNEL_POSITION_FRONT_LEFT:
-				out.push_back(audio::channel_frontLeft);
+				out.pushBack(audio::channel_frontLeft);
 				break;
 			case PA_CHANNEL_POSITION_FRONT_RIGHT:
-				out.push_back(audio::channel_frontRight);
+				out.pushBack(audio::channel_frontRight);
 				break;
 			case PA_CHANNEL_POSITION_REAR_CENTER:
-				out.push_back(audio::channel_rearCenter);
+				out.pushBack(audio::channel_rearCenter);
 				break;
 			case PA_CHANNEL_POSITION_REAR_LEFT:
-				out.push_back(audio::channel_rearLeft);
+				out.pushBack(audio::channel_rearLeft);
 				break;
 			case PA_CHANNEL_POSITION_REAR_RIGHT:
-				out.push_back(audio::channel_rearRight);
+				out.pushBack(audio::channel_rearRight);
 				break;
 			case PA_CHANNEL_POSITION_LFE:
-				out.push_back(audio::channel_lfe);
+				out.pushBack(audio::channel_lfe);
 				break;
 			case PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER:
-				out.push_back(audio::channel_centerLeft);
+				out.pushBack(audio::channel_centerLeft);
 				break;
 			case PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER:
-				out.push_back(audio::channel_centerRight);
+				out.pushBack(audio::channel_centerRight);
 				break;
 			case PA_CHANNEL_POSITION_SIDE_LEFT:
-				out.push_back(audio::channel_topCenterLeft);
+				out.pushBack(audio::channel_topCenterLeft);
 				break;
 			case PA_CHANNEL_POSITION_SIDE_RIGHT:
-				out.push_back(audio::channel_topCenterRight);
+				out.pushBack(audio::channel_topCenterRight);
 				break;
 			case PA_CHANNEL_POSITION_TOP_CENTER:
 			case PA_CHANNEL_POSITION_TOP_FRONT_CENTER:
-				out.push_back(audio::channel_topFrontCenter);
+				out.pushBack(audio::channel_topFrontCenter);
 				break;
 			case PA_CHANNEL_POSITION_TOP_FRONT_LEFT:
-				out.push_back(audio::channel_topFrontLeft);
+				out.pushBack(audio::channel_topFrontLeft);
 				break;
 			case PA_CHANNEL_POSITION_TOP_FRONT_RIGHT:
-				out.push_back(audio::channel_topFrontRight);
+				out.pushBack(audio::channel_topFrontRight);
 				break;
 			case PA_CHANNEL_POSITION_TOP_REAR_LEFT:
-				out.push_back(audio::channel_topRearLeft);
+				out.pushBack(audio::channel_topRearLeft);
 				break;
 			case PA_CHANNEL_POSITION_TOP_REAR_RIGHT:
-				out.push_back(audio::channel_topRearRight);
+				out.pushBack(audio::channel_topRearRight);
 				break;
 			case PA_CHANNEL_POSITION_TOP_REAR_CENTER:
-				out.push_back(audio::channel_topRearCenter);
+				out.pushBack(audio::channel_topRearCenter);
 				break;
-			case PA_CHANNEL_POSITION_AUX0:  out.push_back(audio::channel_aux0);  break;
-			case PA_CHANNEL_POSITION_AUX1:  out.push_back(audio::channel_aux1);  break;
-			case PA_CHANNEL_POSITION_AUX2:  out.push_back(audio::channel_aux2);  break;
-			case PA_CHANNEL_POSITION_AUX3:  out.push_back(audio::channel_aux3);  break;
-			case PA_CHANNEL_POSITION_AUX4:  out.push_back(audio::channel_aux4);  break;
-			case PA_CHANNEL_POSITION_AUX5:  out.push_back(audio::channel_aux5);  break;
-			case PA_CHANNEL_POSITION_AUX6:  out.push_back(audio::channel_aux6);  break;
-			case PA_CHANNEL_POSITION_AUX7:  out.push_back(audio::channel_aux7);  break;
-			case PA_CHANNEL_POSITION_AUX8:  out.push_back(audio::channel_aux8);  break;
-			case PA_CHANNEL_POSITION_AUX9:  out.push_back(audio::channel_aux9);  break;
-			case PA_CHANNEL_POSITION_AUX10: out.push_back(audio::channel_aux10); break;
-			case PA_CHANNEL_POSITION_AUX11: out.push_back(audio::channel_aux11); break;
-			case PA_CHANNEL_POSITION_AUX12: out.push_back(audio::channel_aux12); break;
-			case PA_CHANNEL_POSITION_AUX13: out.push_back(audio::channel_aux13); break;
-			case PA_CHANNEL_POSITION_AUX14: out.push_back(audio::channel_aux14); break;
-			case PA_CHANNEL_POSITION_AUX15: out.push_back(audio::channel_aux15); break;
-			case PA_CHANNEL_POSITION_AUX16: out.push_back(audio::channel_aux16); break;
-			case PA_CHANNEL_POSITION_AUX17: out.push_back(audio::channel_aux17); break;
-			case PA_CHANNEL_POSITION_AUX18: out.push_back(audio::channel_aux18); break;
-			case PA_CHANNEL_POSITION_AUX19: out.push_back(audio::channel_aux19); break;
-			case PA_CHANNEL_POSITION_AUX20: out.push_back(audio::channel_aux20); break;
-			case PA_CHANNEL_POSITION_AUX21: out.push_back(audio::channel_aux21); break;
-			case PA_CHANNEL_POSITION_AUX22: out.push_back(audio::channel_aux22); break;
-			case PA_CHANNEL_POSITION_AUX23: out.push_back(audio::channel_aux23); break;
-			case PA_CHANNEL_POSITION_AUX24: out.push_back(audio::channel_aux24); break;
-			case PA_CHANNEL_POSITION_AUX25: out.push_back(audio::channel_aux25); break;
-			case PA_CHANNEL_POSITION_AUX26: out.push_back(audio::channel_aux26); break;
-			case PA_CHANNEL_POSITION_AUX27: out.push_back(audio::channel_aux27); break;
-			case PA_CHANNEL_POSITION_AUX28: out.push_back(audio::channel_aux28); break;
-			case PA_CHANNEL_POSITION_AUX29: out.push_back(audio::channel_aux29); break;
-			case PA_CHANNEL_POSITION_AUX30: out.push_back(audio::channel_aux30); break;
-			case PA_CHANNEL_POSITION_AUX31: out.push_back(audio::channel_aux31); break;
+			case PA_CHANNEL_POSITION_AUX0:  out.pushBack(audio::channel_aux0);  break;
+			case PA_CHANNEL_POSITION_AUX1:  out.pushBack(audio::channel_aux1);  break;
+			case PA_CHANNEL_POSITION_AUX2:  out.pushBack(audio::channel_aux2);  break;
+			case PA_CHANNEL_POSITION_AUX3:  out.pushBack(audio::channel_aux3);  break;
+			case PA_CHANNEL_POSITION_AUX4:  out.pushBack(audio::channel_aux4);  break;
+			case PA_CHANNEL_POSITION_AUX5:  out.pushBack(audio::channel_aux5);  break;
+			case PA_CHANNEL_POSITION_AUX6:  out.pushBack(audio::channel_aux6);  break;
+			case PA_CHANNEL_POSITION_AUX7:  out.pushBack(audio::channel_aux7);  break;
+			case PA_CHANNEL_POSITION_AUX8:  out.pushBack(audio::channel_aux8);  break;
+			case PA_CHANNEL_POSITION_AUX9:  out.pushBack(audio::channel_aux9);  break;
+			case PA_CHANNEL_POSITION_AUX10: out.pushBack(audio::channel_aux10); break;
+			case PA_CHANNEL_POSITION_AUX11: out.pushBack(audio::channel_aux11); break;
+			case PA_CHANNEL_POSITION_AUX12: out.pushBack(audio::channel_aux12); break;
+			case PA_CHANNEL_POSITION_AUX13: out.pushBack(audio::channel_aux13); break;
+			case PA_CHANNEL_POSITION_AUX14: out.pushBack(audio::channel_aux14); break;
+			case PA_CHANNEL_POSITION_AUX15: out.pushBack(audio::channel_aux15); break;
+			case PA_CHANNEL_POSITION_AUX16: out.pushBack(audio::channel_aux16); break;
+			case PA_CHANNEL_POSITION_AUX17: out.pushBack(audio::channel_aux17); break;
+			case PA_CHANNEL_POSITION_AUX18: out.pushBack(audio::channel_aux18); break;
+			case PA_CHANNEL_POSITION_AUX19: out.pushBack(audio::channel_aux19); break;
+			case PA_CHANNEL_POSITION_AUX20: out.pushBack(audio::channel_aux20); break;
+			case PA_CHANNEL_POSITION_AUX21: out.pushBack(audio::channel_aux21); break;
+			case PA_CHANNEL_POSITION_AUX22: out.pushBack(audio::channel_aux22); break;
+			case PA_CHANNEL_POSITION_AUX23: out.pushBack(audio::channel_aux23); break;
+			case PA_CHANNEL_POSITION_AUX24: out.pushBack(audio::channel_aux24); break;
+			case PA_CHANNEL_POSITION_AUX25: out.pushBack(audio::channel_aux25); break;
+			case PA_CHANNEL_POSITION_AUX26: out.pushBack(audio::channel_aux26); break;
+			case PA_CHANNEL_POSITION_AUX27: out.pushBack(audio::channel_aux27); break;
+			case PA_CHANNEL_POSITION_AUX28: out.pushBack(audio::channel_aux28); break;
+			case PA_CHANNEL_POSITION_AUX29: out.pushBack(audio::channel_aux29); break;
+			case PA_CHANNEL_POSITION_AUX30: out.pushBack(audio::channel_aux30); break;
+			case PA_CHANNEL_POSITION_AUX31: out.pushBack(audio::channel_aux31); break;
 		}
 	}
 	
@@ -207,7 +208,7 @@ static std::vector<audio::channel> getChannelOrderFromPulseChannel(const struct
 }
 // Callback on getting data from pulseaudio:
 static void callbackGetSinkList(pa_context* _contex, const pa_sink_info* _info, int _eol, void* _userdata) {
-	std::vector<audio::orchestra::DeviceInfo>* list = static_cast<std::vector<audio::orchestra::DeviceInfo>*>(_userdata);
+	etk::Vector<audio::orchestra::DeviceInfo>* list = static_cast<etk::Vector<audio::orchestra::DeviceInfo>*>(_userdata);
 	// If eol is set to a positive number, you're at the end of the list
 	if (_eol > 0) {
 		return;
@@ -217,17 +218,17 @@ static void callbackGetSinkList(pa_context* _contex, const pa_sink_info* _info,
 	info.input = false;
 	info.name = _info->name;
 	info.desc = _info->description;
-	info.sampleRates.push_back(_info->sample_spec.rate);
-	info.nativeFormats.push_back(getFormatFromPulseFormat(_info->sample_spec.format));
+	info.sampleRates.pushBack(_info->sample_spec.rate);
+	info.nativeFormats.pushBack(getFormatFromPulseFormat(_info->sample_spec.format));
 	info.channels = getChannelOrderFromPulseChannel(_info->channel_map);
 	ATA_VERBOSE("plop=" << _info->index << " " << _info->name);
 	//ATA_DEBUG("          ports=" << _info->n_ports);
-	list->push_back(info);
+	list->pushBack(info);
 }
 
 // allback to get data from pulseaudio:
 static void callbackGetSourceList(pa_context* _contex, const pa_source_info* _info, int _eol, void* _userdata) {
-	std::vector<audio::orchestra::DeviceInfo>* list = static_cast<std::vector<audio::orchestra::DeviceInfo>*>(_userdata);
+	etk::Vector<audio::orchestra::DeviceInfo>* list = static_cast<etk::Vector<audio::orchestra::DeviceInfo>*>(_userdata);
 	if (_eol > 0) {
 		return;
 	}
@@ -236,18 +237,18 @@ static void callbackGetSourceList(pa_context* _contex, const pa_source_info* _in
 	info.input = true;
 	info.name = _info->name;
 	info.desc = _info->description;
-	info.sampleRates.push_back(_info->sample_spec.rate);
-	info.nativeFormats.push_back(getFormatFromPulseFormat(_info->sample_spec.format));
+	info.sampleRates.pushBack(_info->sample_spec.rate);
+	info.nativeFormats.pushBack(getFormatFromPulseFormat(_info->sample_spec.format));
 	info.channels = getChannelOrderFromPulseChannel(_info->channel_map);
 	ATA_VERBOSE("plop=" << _info->index << " " << _info->name);
-	list->push_back(info);
+	list->pushBack(info);
 }
 
 // to not update all the time ...
-static std::vector<audio::orchestra::DeviceInfo> pulseAudioListOfDevice;
+static etk::Vector<audio::orchestra::DeviceInfo> pulseAudioListOfDevice;
 static audio::Time pulseAudioListOfDeviceTime;
 
-std::vector<audio::orchestra::DeviceInfo> audio::orchestra::api::pulse::getDeviceList() {
+etk::Vector<audio::orchestra::DeviceInfo> audio::orchestra::api::pulse::getDeviceList() {
 	audio::Duration delta = audio::Time::now() - pulseAudioListOfDeviceTime;
 	if (delta < audio::Duration(30,0)) {
 		return pulseAudioListOfDevice;
@@ -258,7 +259,7 @@ std::vector<audio::orchestra::DeviceInfo> audio::orchestra::api::pulse::getDevic
 	pa_operation* pulseAudioOperation;
 	pa_context* pulseAudioContex;
 	pa_context_flags_t pulseAudioFlags = PA_CONTEXT_NOAUTOSPAWN;
-	std::vector<audio::orchestra::DeviceInfo>& out = pulseAudioListOfDevice;
+	etk::Vector<audio::orchestra::DeviceInfo>& out = pulseAudioListOfDevice;
 	out.clear();
 	// We'll need these state variables to keep track of our requests
 	int state = 0;
@@ -276,7 +277,7 @@ std::vector<audio::orchestra::DeviceInfo> audio::orchestra::api::pulse::getDevic
 		// We can't do anything until PA is ready, so just iterate the mainloop
 		// and continue
 		if (pulseAudioReady == 0) {
-			pa_mainloop_iterate(pulseAudioMainLoop, 1, nullptr);
+			pa_mainloop_iterate(pulseAudioMainLoop, 1, null);
 			continue;
 		}
 		// We couldn't get a connection to the server, so exit out
@@ -330,7 +331,7 @@ std::vector<audio::orchestra::DeviceInfo> audio::orchestra::api::pulse::getDevic
 		}
 		// Iterate the main loop ..
 		if (playLoop == true) {
-			pa_mainloop_iterate(pulseAudioMainLoop, 1, nullptr);
+			pa_mainloop_iterate(pulseAudioMainLoop, 1, null);
 		}
 	}
 	// TODO: need to do it better ...

audio/orchestra/api/PulseDeviceList.hpp

@@ -7,14 +7,14 @@
 #pragma once
 #ifdef ORCHESTRA_BUILD_PULSE
 
-#include <etk/types.h>
-#include <audio/orchestra/DeviceInfo.h>
+#include <etk/types.hpp>
+#include <audio/orchestra/DeviceInfo.hpp>
 
 namespace audio {
 	namespace orchestra {
 		namespace api {
 			namespace pulse {
-				std::vector<audio::orchestra::DeviceInfo> getDeviceList();
+				etk::Vector<audio::orchestra::DeviceInfo> getDeviceList();
 			}
 		}
 	}

audio/orchestra/base.h

@@ -1,25 +0,0 @@
-/** @file
- * @author Edouard DUPIN 
- * @copyright 2011, Edouard DUPIN, all right reserved
- * @license APACHE v2.0 (see license file)
- * @fork from RTAudio
- */
-#pragma once
-
-#include <thread>
-#include <condition_variable>
-#include <mutex>
-#include <chrono>
-#include <functional>
-#include <ememory/memory.h>
-#include <audio/channel.h>
-#include <audio/format.h>
-#include <audio/orchestra/error.h>
-#include <audio/orchestra/status.h>
-#include <audio/orchestra/Flags.h>
-
-#include <audio/orchestra/CallbackInfo.h>
-#include <audio/orchestra/DeviceInfo.h>
-#include <audio/orchestra/StreamOptions.h>
-#include <audio/orchestra/StreamParameters.h>
-

audio/orchestra/base.hpp

@@ -0,0 +1,25 @@
+/** @file
+ * @author Edouard DUPIN 
+ * @copyright 2011, Edouard DUPIN, all right reserved
+ * @license APACHE v2.0 (see license file)
+ * @fork from RTAudio
+ */
+#pragma once
+
+#include <ethread/Thread.hpp>
+#include <ethread/Semaphore.hpp>
+#include <ethread/Mutex.hpp>
+#include <echrono/Steady.hpp>
+#include <etk/Function.hpp>
+#include <ememory/memory.hpp>
+#include <audio/channel.hpp>
+#include <audio/format.hpp>
+#include <audio/orchestra/error.hpp>
+#include <audio/orchestra/status.hpp>
+#include <audio/orchestra/Flags.hpp>
+
+#include <audio/orchestra/CallbackInfo.hpp>
+#include <audio/orchestra/DeviceInfo.hpp>
+#include <audio/orchestra/StreamOptions.hpp>
+#include <audio/orchestra/StreamParameters.hpp>
+

audio/orchestra/debug.cpp

@@ -5,7 +5,7 @@
  * @fork from RTAudio
  */
 
-#include <audio/orchestra/debug.h>
+#include <audio/orchestra/debug.hpp>
 
 int32_t audio::orchestra::getLogId() {
 	static int32_t g_val = elog::registerInstance("audio-orchestra");

audio/orchestra/debug.hpp

@@ -6,7 +6,7 @@
  */
 #pragma once
 
-#include <elog/log.h>
+#include <elog/log.hpp>
 
 namespace audio {
 	namespace orchestra {

audio/orchestra/error.cpp

@@ -5,5 +5,5 @@
  * @fork from RTAudio
  */
 
-#include <audio/orchestra/error.h>
-#include <audio/orchestra/debug.h>
+#include <audio/orchestra/error.hpp>
+#include <audio/orchestra/debug.hpp>

audio/orchestra/error.hpp

@@ -6,7 +6,7 @@
  */
 #pragma once
 
-#include <etk/types.h>
+#include <etk/types.hpp>
 
 namespace audio {
 	namespace orchestra {

audio/orchestra/mode.cpp

@@ -5,8 +5,8 @@
  * @fork from RTAudio
  */
 
-#include <audio/orchestra/mode.h>
-#include <audio/orchestra/debug.h>
+#include <audio/orchestra/mode.hpp>
+#include <audio/orchestra/debug.hpp>
 
 int32_t audio::orchestra::modeToIdTable(enum mode _mode) {
 	switch (_mode) {
@@ -20,7 +20,7 @@ int32_t audio::orchestra::modeToIdTable(enum mode _mode) {
 	return 0;
 }
 
-std::ostream& audio::operator <<(std::ostream& _os, enum audio::orchestra::mode _obj) {
+etk::Stream& audio::operator <<(etk::Stream& _os, enum audio::orchestra::mode _obj) {
 	switch (_obj) {
 		case audio::orchestra::mode_unknow:
 			_os << "unknow";

audio/orchestra/mode.hpp

@@ -6,7 +6,8 @@
  */
 #pragma once
 
-#include <etk/types.h>
+#include <etk/types.hpp>
+#include <etk/Stream.hpp>
 
 namespace audio {
 	namespace orchestra {
@@ -18,6 +19,6 @@ namespace audio {
 		};
 		int32_t modeToIdTable(enum mode _mode);
 	}
-	std::ostream& operator <<(std::ostream& _os, enum audio::orchestra::mode _obj);
+	etk::Stream& operator <<(etk::Stream& _os, enum audio::orchestra::mode _obj);
 }
 

audio/orchestra/state.hpp

@@ -6,7 +6,7 @@
  */
 #pragma once
 
-#include <etk/types.h>
+#include <etk/types.hpp>
 
 namespace audio {
 	namespace orchestra {

audio/orchestra/status.cpp

@@ -5,28 +5,28 @@
  * @fork from RTAudio
  */
 
-#include <audio/orchestra/status.h>
-#include <audio/orchestra/debug.h>
+#include <audio/orchestra/status.hpp>
+#include <audio/orchestra/debug.hpp>
 static const char* listValue[] = {
 	"ok",
 	"overflow",
 	"underflow"
 };
 
-std::ostream& audio::orchestra::operator <<(std::ostream& _os, enum audio::orchestra::status _obj) {
+etk::Stream& audio::orchestra::operator <<(etk::Stream& _os, enum audio::orchestra::status _obj) {
 	_os << listValue[int32_t(_obj)];
 	return _os;
 }
 
-std::ostream& audio::orchestra::operator <<(std::ostream& _os, const std::vector<enum audio::orchestra::status>& _obj) {
-	_os << std::string("{");
+etk::Stream& audio::orchestra::operator <<(etk::Stream& _os, const etk::Vector<enum audio::orchestra::status>& _obj) {
+	_os << etk::String("{");
 	for (size_t iii=0; iii<_obj.size(); ++iii) {
 		if (iii!=0) {
-			_os << std::string(";");
+			_os << etk::String(";");
 		}
 		_os << _obj[iii];
 	}
-	_os << std::string("}");
+	_os << etk::String("}");
 	return _os;
 }
 

audio/orchestra/status.hpp

@@ -6,7 +6,8 @@
  */
 #pragma once
 
-#include <etk/types.h>
+#include <etk/types.hpp>
+#include <etk/Vector.hpp>
 
 namespace audio {
 	namespace orchestra {
@@ -15,8 +16,8 @@ namespace audio {
 			overflow, //!< Internal buffer has more data than they can accept
 			underflow //!< The internal buffer is empty
 		};
-		std::ostream& operator <<(std::ostream& _os, enum audio::orchestra::status _obj);
-		std::ostream& operator <<(std::ostream& _os, const std::vector<enum audio::orchestra::status>& _obj);
+		etk::Stream& operator <<(etk::Stream& _os, enum audio::orchestra::status _obj);
+		etk::Stream& operator <<(etk::Stream& _os, const etk::Vector<enum audio::orchestra::status>& _obj);
 	}
 }
 

audio/orchestra/type.cpp

@@ -5,21 +5,17 @@
  * @fork from RTAudio
  */
 
-#include <audio/orchestra/type.h>
-#include <audio/orchestra/debug.h>
-#include <iostream>
-#include <cstdlib>
-#include <cstring>
-#include <climits>
+#include <audio/orchestra/type.hpp>
+#include <audio/orchestra/debug.hpp>
 
-const std::string audio::orchestra::typeUndefined = "undefined";
-const std::string audio::orchestra::typeAlsa = "alsa";
-const std::string audio::orchestra::typePulse = "pulse";
-const std::string audio::orchestra::typeOss = "oss";
-const std::string audio::orchestra::typeJack = "jack";
-const std::string audio::orchestra::typeCoreOSX = "coreOSX";
-const std::string audio::orchestra::typeCoreIOS = "coreIOS";
-const std::string audio::orchestra::typeAsio = "asio";
-const std::string audio::orchestra::typeDs = "ds";
-const std::string audio::orchestra::typeJava = "java";
-const std::string audio::orchestra::typeDummy = "dummy";
+const etk::String audio::orchestra::typeUndefined = "undefined";
+const etk::String audio::orchestra::typeAlsa = "alsa";
+const etk::String audio::orchestra::typePulse = "pulse";
+const etk::String audio::orchestra::typeOss = "oss";
+const etk::String audio::orchestra::typeJack = "jack";
+const etk::String audio::orchestra::typeCoreOSX = "coreOSX";
+const etk::String audio::orchestra::typeCoreIOS = "coreIOS";
+const etk::String audio::orchestra::typeAsio = "asio";
+const etk::String audio::orchestra::typeDs = "ds";
+const etk::String audio::orchestra::typeJava = "java";
+const etk::String audio::orchestra::typeDummy = "dummy";

audio/orchestra/type.h

@@ -1,30 +0,0 @@
-/** @file
- * @author Edouard DUPIN 
- * @copyright 2011, Edouard DUPIN, all right reserved
- * @license APACHE v2.0 (see license file)
- * @fork from RTAudio
- */
-#pragma once
-
-#include <etk/types.h>
-#include <etk/stdTools.h>
-
-namespace audio {
-	namespace orchestra {
-		/**
-		 * @brief Audio API specifier arguments.
-		 */
-		extern const std::string typeUndefined; //!< Error API.
-		extern const std::string typeAlsa; //!< LINUX The Advanced Linux Sound Architecture.
-		extern const std::string typePulse; //!< LINUX The Linux PulseAudio.
-		extern const std::string typeOss; //!< LINUX The Linux Open Sound System.
-		extern const std::string typeJack; //!< UNIX The Jack Low-Latency Audio Server.
-		extern const std::string typeCoreOSX; //!< Macintosh OSX Core Audio.
-		extern const std::string typeCoreIOS; //!< Macintosh iOS Core Audio.
-		extern const std::string typeAsio; //!< WINDOWS The Steinberg Audio Stream I/O.
-		extern const std::string typeDs; //!< WINDOWS The Microsoft Direct Sound.
-		extern const std::string typeJava; //!< ANDROID Interface.
-		extern const std::string typeDummy; //!< Empty wrapper (non-functional).
-	}
-}
-

audio/orchestra/type.hpp

@@ -0,0 +1,30 @@
+/** @file
+ * @author Edouard DUPIN 
+ * @copyright 2011, Edouard DUPIN, all right reserved
+ * @license APACHE v2.0 (see license file)
+ * @fork from RTAudio
+ */
+#pragma once
+
+#include <etk/types.hpp>
+#include <etk/stdTools.hpp>
+
+namespace audio {
+	namespace orchestra {
+		/**
+		 * @brief Audio API specifier arguments.
+		 */
+		extern const etk::String typeUndefined; //!< Error API.
+		extern const etk::String typeAlsa; //!< LINUX The Advanced Linux Sound Architecture.
+		extern const etk::String typePulse; //!< LINUX The Linux PulseAudio.
+		extern const etk::String typeOss; //!< LINUX The Linux Open Sound System.
+		extern const etk::String typeJack; //!< UNIX The Jack Low-Latency Audio Server.
+		extern const etk::String typeCoreOSX; //!< Macintosh OSX Core Audio.
+		extern const etk::String typeCoreIOS; //!< Macintosh iOS Core Audio.
+		extern const etk::String typeAsio; //!< WINDOWS The Steinberg Audio Stream I/O.
+		extern const etk::String typeDs; //!< WINDOWS The Microsoft Direct Sound.
+		extern const etk::String typeJava; //!< ANDROID Interface.
+		extern const etk::String typeDummy; //!< Empty wrapper (non-functional).
+	}
+}
+

authors.txt

@@ -0,0 +1 @@
+MR Edouard DUPIN <yui.heero@gmail.com>

catkin/CMakeLists.txt

@@ -1,91 +0,0 @@
-cmake_minimum_required(VERSION 2.8.3)
-project(audio_orchestra)
-
-set(CMAKE_VERBOSE_MAKEFILE ON)
-
-## Find catkin macros and libraries
-## if COMPONENTS list like find_package(catkin REQUIRED COMPONENTS xyz)
-## is used, also find other catkin packages
-find_package(catkin REQUIRED COMPONENTS
-  etk
-  audio
-  )
-
-find_package(ALSA REQUIRED)
-
-###################################
-## catkin specific configuration ##
-###################################
-## The catkin_package macro generates cmake config files for your package
-## Declare things to be passed to dependent projects
-## INCLUDE_DIRS: uncomment this if you package contains header files
-## LIBRARIES: libraries you create in this project that dependent projects also need
-## CATKIN_DEPENDS: catkin_packages dependent projects also need
-## DEPENDS: system dependencies of this project that dependent projects also need
-catkin_package(
-  INCLUDE_DIRS ../
-  LIBRARIES ${PROJECT_NAME}
-  CATKIN_DEPENDS etk audio
-  DEPENDS system_lib
-)
-
-###########
-## Build ##
-###########
-
-## Specify additional locations of header files
-## Your package locations should be listed before other locations
-include_directories(
-  ..
-  ${catkin_INCLUDE_DIRS}
-)
-
-## Declare a cpp library
-add_library(${PROJECT_NAME}
-  ../audio/orchestra/debug.cpp
-  ../audio/orchestra/status.cpp
-  ../audio/orchestra/type.cpp
-  ../audio/orchestra/mode.cpp
-  ../audio/orchestra/state.cpp
-  ../audio/orchestra/error.cpp
-  ../audio/orchestra/base.cpp
-  ../audio/orchestra/Interface.cpp
-  ../audio/orchestra/Flags.cpp
-  ../audio/orchestra/Api.cpp
-  ../audio/orchestra/DeviceInfo.cpp
-  ../audio/orchestra/StreamOptions.cpp
-  ../audio/orchestra/api/Dummy.cpp
-  ../audio/orchestra/api/Alsa.cpp
-  ../audio/orchestra/api/Jack.cpp
-  ../audio/orchestra/api/Pulse.cpp
-  ../audio/orchestra/api/Oss.cpp
-)
-
-add_definitions(-D__LINUX_ALSA__)
-add_definitions(-D__DUMMY__)
-
-## Add cmake target dependencies of the executable/library
-## as an example, message headers may need to be generated before nodes
-#add_dependencies(${PROJECT_NAME} test_perfo_core_generate_messages_cpp)
-
-## Specify libraries to link a library or executable target against
-target_link_libraries(${PROJECT_NAME}
-  ${ALSA_LIBRARIES}
-  ${catkin_LIBRARIES}
-)
-
-#############
-## Install ##
-#############
-
-## Mark executables and/or libraries for installation
-install(TARGETS ${PROJECT_NAME}
-  ARCHIVE DESTINATION ${CATKIN_PACKAGE_LIB_DESTINATION}
-  LIBRARY DESTINATION ${CATKIN_PACKAGE_LIB_DESTINATION}
-)
-
-## Mark cpp header files for installation
-install(DIRECTORY ../audio/orchestra/
-  DESTINATION ${CATKIN_PACKAGE_INCLUDE_DESTINATION}
-  FILES_MATCHING PATTERN "*.h"
-)

catkin/package.xml

@@ -1,13 +0,0 @@
-<?xml version="1.0"?>
-<package>
-  <name>audio_orchestra</name>
-  <version>0.3.0</version>
-  <description>Ewol RTAudio fork</description>
-  <maintainer email="yui.heero@gmail.com">Edouard DUPIN</maintainer>
-  <license>Apache-2.0</license>
-  <build_depend>etk</build_depend>
-  <build_depend>audio</build_depend>
-  <buildtool_depend>catkin</buildtool_depend>
-  <run_depend>etk</run_depend>
-  <run_depend>audio</run_depend>
-</package>

doc/build.md

@@ -0,0 +1,83 @@
+Build lib & build sample                           {#audio_orchestra_build}
+========================
+
+@tableofcontents
+
+Download:                                          {#audio_orchestra_build_download}
+=========
+
+ege use some tools to manage source and build it:
+
+need google repo:                                  {#audio_orchestra_build_download_repo}
+-----------------
+
+see: http://source.android.com/source/downloading.html#installing-repo
+
+On all platform:
+```{.sh}
+	mkdir ~/.bin
+	PATH=~/.bin:$PATH
+	curl https://storage.googleapis.com/git-repo-downloads/repo > ~/.bin/repo
+	chmod a+x ~/.bin/repo
+```
+
+On ubuntu
+```{.sh}
+	sudo apt-get install repo
+```
+
+On archlinux
+```{.sh}
+	sudo pacman -S repo
+```
+
+lutin (build-system):                              {#audio_orchestra_build_download_lutin}
+---------------------
+
+```{.sh}
+	pip install lutin --user
+	# optionnal dependency of lutin (manage image changing size for application release)
+	pip install pillow --user
+```
+
+
+dependency:                                        {#audio_orchestra_build_download_dependency}
+-----------
+
+```{.sh}
+	mkdir -p WORKING_DIRECTORY/framework
+	cd WORKING_DIRECTORY/framework
+	repo init -u git://github.com/atria-soft/manifest.git
+	repo sync -j8
+	cd ../..
+```
+
+sources:                                           {#audio_orchestra_build_download_sources}
+--------
+
+They are already download in the repo manifest in:
+
+```{.sh}
+	cd WORKING_DIRECTORY/framework/atria-soft/audio_orchestra
+```
+
+Build:                                             {#audio_orchestra_build_build}
+======
+
+you must stay in zour working directory...
+```{.sh}
+	cd WORKING_DIRECTORY
+```
+
+library:                                           {#audio_orchestra_build_build_library}
+--------
+
+```{.sh}
+	lutin -mdebug audio-orchestra
+```
+
+Sample:                                            {#audio_orchestra_build_build_sample}
+-------
+
+No smaple availlable for now ...
+

doc/mainpage.md

@@ -0,0 +1,50 @@
+AUDIO-ORCHESTRA library                                {#mainpage}
+=======================
+
+@tableofcontents
+
+What is AUDIO-ORCHESTRA:                               {#audio_orchestra_mainpage_what}
+========================
+
+AUDIO-ORCHESTRA, is a fork of RTAudio lib (with port for Android, and IOs).
+
+This is a cross API of Audio wrapping the Hardware.
+
+This library is not friendy usable, use audio-river to have a correct and simple multiple-flow API
+
+What it does:                               {#audio_orchestra_mainpage_what_it_does}
+-------------
+
+  - Open a strem audio in Input or output
+  - synchronise 2 flow
+  - open on some platform: Android, Linux, MacOs, Ios, Windows
+
+AUDIO-ORCHESTRA is dependent of the STL (compatible with MacOs stl (CXX))
+
+
+What languages are supported?                    {#audio_orchestra_mainpage_language}
+=============================
+
+AUDIO-ORCHESTRA is written in C++.
+
+
+Are there any licensing restrictions?            {#audio_orchestra_mainpage_license_restriction}
+=====================================
+
+AUDIO-ORCHESTRA is **FREE software** and _all sub-library are FREE and staticly linkable !!!_
+
+
+License (MIT)                             {#audio_orchestra_mainpage_license}
+=============
+
+Copyright AUDIO-ORCHESTRA Edouard DUPIN
+
+MIT ...
+
+
+Other pages                              {#audio_orchestra_mainpage_sub_page}
+===========
+
+  - @ref audio_orchestra_build
+  - [**ewol coding style**](http://atria-soft.github.io/ewol/ewol_coding_style.html)
+

doxy_audio-orchestra.py

@@ -0,0 +1,32 @@
+#!/usr/bin/python
+import os
+import doxy.module as module
+import doxy.debug as debug
+import doxy.tools as tools
+
+def create(target, module_name):
+	my_module = module.Module(__file__, module_name)
+	my_module.set_version("version.txt")
+	my_module.set_title("Orchestra: audio interface wrapper")
+	my_module.set_website("http://musicdsp.github.io/" + module_name)
+	my_module.set_website_sources("http://github.com/musicdsp/" + module_name)
+	my_module.add_path([
+	    "audio",
+	    "doc"
+	    ])
+	my_module.add_depend([
+	    'etk',
+	    'audio',
+	    ])
+	my_module.add_exclude_symbols([
+	    '*operator<<*',
+	    ])
+	my_module.add_exclude_file([
+	    'debug.h',
+	    ])
+	my_module.add_file_patterns([
+	    '*.h',
+	    '*.md',
+	    ])
+	
+	return my_module

lutin_audio-orchestra.py

@@ -1,7 +1,6 @@
 #!/usr/bin/python
-import lutin.module as module
 import lutin.tools as tools
-import lutin.debug as debug
+import realog.debug as debug
 
 
 def get_type():
@@ -20,13 +19,12 @@ def get_compagny_name():
 	return "atria-soft"
 
 def get_maintainer():
-	return ["Mr DUPIN Edouard <yui.heero@gmail.com>"]
+	return "authors.txt"
 
 def get_version():
-	return [0,3,0]
+	return "version.txt"
 
-def create(target, module_name):
-	my_module = module.Module(__file__, module_name, get_type())
+def configure(target, my_module):
 	my_module.add_src_file([
 		'audio/orchestra/debug.cpp',
 		'audio/orchestra/status.cpp',
@@ -43,67 +41,70 @@ def create(target, module_name):
 		'audio/orchestra/api/Dummy.cpp'
 		])
 	my_module.add_header_file([
-		'audio/orchestra/debug.h',
-		'audio/orchestra/status.h',
-		'audio/orchestra/type.h',
-		'audio/orchestra/mode.h',
-		'audio/orchestra/state.h',
-		'audio/orchestra/error.h',
-		'audio/orchestra/base.h',
-		'audio/orchestra/Interface.h',
-		'audio/orchestra/Flags.h',
-		'audio/orchestra/Api.h',
-		'audio/orchestra/DeviceInfo.h',
-		'audio/orchestra/StreamOptions.h',
-		'audio/orchestra/CallbackInfo.h',
-		'audio/orchestra/StreamParameters.h'
+		'audio/orchestra/debug.hpp',
+		'audio/orchestra/status.hpp',
+		'audio/orchestra/type.hpp',
+		'audio/orchestra/mode.hpp',
+		'audio/orchestra/state.hpp',
+		'audio/orchestra/error.hpp',
+		'audio/orchestra/base.hpp',
+		'audio/orchestra/Interface.hpp',
+		'audio/orchestra/Flags.hpp',
+		'audio/orchestra/Api.hpp',
+		'audio/orchestra/DeviceInfo.hpp',
+		'audio/orchestra/StreamOptions.hpp',
+		'audio/orchestra/CallbackInfo.hpp',
+		'audio/orchestra/StreamParameters.hpp'
 		])
-	my_module.add_module_depend(['audio', 'etk'])
+	my_module.add_depend([
+	    'audio',
+	    'etk'
+	    ])
 	# add all the time the dummy interface
-	my_module.add_export_flag('c++', ['-DORCHESTRA_BUILD_DUMMY'])
+	my_module.add_flag('c++', ['-DORCHESTRA_BUILD_DUMMY'], export=True)
 	# TODO : Add a FILE interface:
 	
-	if target.config_based_on=="Windows":
+	if "Windows" in target.get_type():
 		my_module.add_src_file([
-			'audio/orchestra/api/Asio.cpp',
-			'audio/orchestra/api/Ds.cpp',
-			])
+		    'audio/orchestra/api/Asio.cpp',
+		    'audio/orchestra/api/Ds.cpp',
+		    ])
 		# load optionnal API:
-		my_module.add_optionnal_module_depend('asio', ["c++", "-DORCHESTRA_BUILD_ASIO"])
-		my_module.add_optionnal_module_depend('ds', ["c++", "-DORCHESTRA_BUILD_DS"])
-		my_module.add_optionnal_module_depend('wasapi', ["c++", "-DORCHESTRA_BUILD_WASAPI"])
-	elif target.config_based_on=="Linux":
+		my_module.add_optionnal_depend('asio', ["c++", "-DORCHESTRA_BUILD_ASIO"])
+		my_module.add_optionnal_depend('ds', ["c++", "-DORCHESTRA_BUILD_DS"])
+		my_module.add_optionnal_depend('wasapi', ["c++", "-DORCHESTRA_BUILD_WASAPI"])
+	elif "Linux" in target.get_type():
 		my_module.add_src_file([
-			'audio/orchestra/api/Alsa.cpp',
-			'audio/orchestra/api/Jack.cpp',
-			'audio/orchestra/api/Pulse.cpp',
-			'audio/orchestra/api/PulseDeviceList.cpp'
-			])
-		my_module.add_optionnal_module_depend('alsa', ["c++", "-DORCHESTRA_BUILD_ALSA"])
-		my_module.add_optionnal_module_depend('jack', ["c++", "-DORCHESTRA_BUILD_JACK"])
-		my_module.add_optionnal_module_depend('pulse', ["c++", "-DORCHESTRA_BUILD_PULSE"])
-	elif target.config_based_on=="MacOs":
+		    'audio/orchestra/api/Alsa.cpp',
+		    'audio/orchestra/api/Jack.cpp',
+		    'audio/orchestra/api/Pulse.cpp',
+		    'audio/orchestra/api/PulseDeviceList.cpp'
+		    ])
+		my_module.add_optionnal_depend('alsa', ["c++", "-DORCHESTRA_BUILD_ALSA"])
+		my_module.add_optionnal_depend('jack', ["c++", "-DORCHESTRA_BUILD_JACK"])
+		my_module.add_optionnal_depend('pulse', ["c++", "-DORCHESTRA_BUILD_PULSE"])
+	elif "MacOs" in target.get_type():
 		my_module.add_src_file([
-							   'audio/orchestra/api/Core.cpp'
-							   ])
+		    'audio/orchestra/api/Core.cpp'
+		    ])
 		# MacOsX core
-		my_module.add_optionnal_module_depend('CoreAudio', ["c++", "-DORCHESTRA_BUILD_MACOSX_CORE"])
-	elif target.config_based_on=="IOs":
+		my_module.add_optionnal_depend('CoreAudio', ["c++", "-DORCHESTRA_BUILD_MACOSX_CORE"])
+	elif "IOs" in target.get_type():
 		my_module.add_src_file('audio/orchestra/api/CoreIos.mm')
 		# IOsX core
-		my_module.add_optionnal_module_depend('CoreAudio', ["c++", "-DORCHESTRA_BUILD_IOS_CORE"])
-	elif target.config_based_on=="Android":
+		my_module.add_optionnal_depend('CoreAudio', ["c++", "-DORCHESTRA_BUILD_IOS_CORE"])
+	elif "Android" in target.get_type():
 		my_module.add_src_file('android/org/musicdsp/orchestra/OrchestraConstants.java')
 		my_module.add_src_file('android/org/musicdsp/orchestra/OrchestraManagerCallback.java')
 		my_module.add_src_file('android/org/musicdsp/orchestra/OrchestraNative.java')
 		my_module.add_src_file('android/org/musicdsp/orchestra/OrchestraInterfaceInput.java')
 		my_module.add_src_file('android/org/musicdsp/orchestra/OrchestraInterfaceOutput.java')
 		my_module.add_src_file('android/org/musicdsp/orchestra/OrchestraManager.java')
-		# create inter language interface
+		# create inter language interfacef
 		my_module.add_src_file('org.musicdsp.orchestra.OrchestraConstants.javah')
-		my_module.add_path(tools.get_current_path(__file__) + '/android/', type='java')
-		my_module.add_module_depend(['SDK', 'jvm-basics', 'ejson'])
-		my_module.add_export_flag('c++', ['-DORCHESTRA_BUILD_JAVA'])
+		my_module.add_path('android', type='java')
+		my_module.add_depend(['SDK', 'jvm-basics', 'ejson'])
+		my_module.add_flag('c++', ['-DORCHESTRA_BUILD_JAVA'], export=True)
 		
 		my_module.add_src_file('audio/orchestra/api/Android.cpp')
 		my_module.add_src_file('audio/orchestra/api/AndroidNativeInterface.cpp')
@@ -112,9 +113,9 @@ def create(target, module_name):
 	else:
 		debug.warning("unknow target for audio_orchestra : " + target.name);
 	
-	my_module.add_path(tools.get_current_path(__file__))
+	my_module.add_path(".")
 	
-	return my_module
+	return True
 
 
 
@@ -124,13 +125,13 @@ def create(target, module_name):
 ##
 ##################################################################
 def tool_generate_add_java_section_in_class(target, module, package_name):
-	module.pkg_add("GENERATE_SECTION__IMPORT", [
+	module.add_pkg("GENERATE_SECTION__IMPORT", [
 		"import org.musicdsp.orchestra.OrchestraManager;"
 		])
-	module.pkg_add("GENERATE_SECTION__DECLARE", [
+	module.add_pkg("GENERATE_SECTION__DECLARE", [
 		"private OrchestraManager m_audioManagerHandle;"
 		])
-	module.pkg_add("GENERATE_SECTION__CONSTRUCTOR", [
+	module.add_pkg("GENERATE_SECTION__CONSTRUCTOR", [
 		"// load audio maneger if it does not work, it is not critical ...",
 		"try {",
 		"	m_audioManagerHandle = new OrchestraManager();",
@@ -138,37 +139,37 @@ def tool_generate_add_java_section_in_class(target, module, package_name):
 		"	Log.e(\"" + package_name + "\", \"Can not load Audio interface (maybe not really needed) :\" + e);",
 		"}"
 		])
-	module.pkg_add("GENERATE_SECTION__ON_CREATE", [
+	module.add_pkg("GENERATE_SECTION__ON_CREATE", [
 		"if (m_audioManagerHandle != null) {",
 		"	m_audioManagerHandle.onCreate();",
 		"}"
 		])
-	module.pkg_add("GENERATE_SECTION__ON_START", [
+	module.add_pkg("GENERATE_SECTION__ON_START", [
 		"if (m_audioManagerHandle != null) {",
 		"	m_audioManagerHandle.onStart();",
 		"}"
 		])
-	module.pkg_add("GENERATE_SECTION__ON_RESTART", [
+	module.add_pkg("GENERATE_SECTION__ON_RESTART", [
 		"if (m_audioManagerHandle != null) {",
 		"	m_audioManagerHandle.onRestart();",
 		"}"
 		])
-	module.pkg_add("GENERATE_SECTION__ON_RESUME", [
+	module.add_pkg("GENERATE_SECTION__ON_RESUME", [
 		"if (m_audioManagerHandle != null) {",
 		"	m_audioManagerHandle.onResume();",
 		"}"
 		])
-	module.pkg_add("GENERATE_SECTION__ON_PAUSE", [
+	module.add_pkg("GENERATE_SECTION__ON_PAUSE", [
 		"if (m_audioManagerHandle != null) {",
 		"	m_audioManagerHandle.onPause();",
 		"}"
 		])
-	module.pkg_add("GENERATE_SECTION__ON_STOP", [
+	module.add_pkg("GENERATE_SECTION__ON_STOP", [
 		"if (m_audioManagerHandle != null) {",
 		"	m_audioManagerHandle.onStop();",
 		"}"
 		])
-	module.pkg_add("GENERATE_SECTION__ON_DESTROY", [
+	module.add_pkg("GENERATE_SECTION__ON_DESTROY", [
 		"// Destroy the AdView.",
 		"if (m_audioManagerHandle != null) {",
 		"	m_audioManagerHandle.onDestroy();",

tools/lutin_orchestra-in.py

@@ -1,7 +1,6 @@
 #!/usr/bin/python
-import lutin.module as module
+import realog.debug as debug
 import lutin.tools as tools
-import lutin.debug as debug
 
 
 def get_type():
@@ -25,13 +24,14 @@ def get_compagny_name():
 def get_maintainer():
 	return ["Mr DUPIN Edouard <yui.heero@gmail.com>"]
 
-def create(target, module_name):
-	my_module = module.Module(__file__, module_name, get_type())
-	
+def configure(target, my_module):
 	my_module.add_src_file([
-		'orchestra-in.cpp'
-		])
-	my_module.add_module_depend(['audio-orchestra', 'test-debug'])
-	return my_module
+	    'orchestra-in.cpp'
+	    ])
+	my_module.add_depend([
+	    'audio-orchestra',
+	    'test-debug'
+	    ])
+	return True
 
 

tools/lutin_orchestra-list.py

@@ -1,7 +1,6 @@
 #!/usr/bin/python
-import lutin.module as module
+import realog.debug as debug
 import lutin.tools as tools
-import lutin.debug as debug
 
 
 def get_type():
@@ -25,13 +24,14 @@ def get_compagny_name():
 def get_maintainer():
 	return ["Mr DUPIN Edouard <yui.heero@gmail.com>"]
 
-def create(target, module_name):
-	my_module = module.Module(__file__, module_name, get_type())
-	
+def configure(target, my_module):
 	my_module.add_src_file([
-		'orchestra-list.cpp'
-		])
-	my_module.add_module_depend(['audio-orchestra', 'test-debug'])
-	return my_module
+	    'orchestra-list.cpp'
+	    ])
+	my_module.add_depend([
+	    'audio-orchestra',
+	    'test-debug'
+	    ])
+	return True
 
 

tools/lutin_orchestra-out.py

@@ -1,7 +1,6 @@
 #!/usr/bin/python
-import lutin.module as module
+import realog.debug as debug
 import lutin.tools as tools
-import lutin.debug as debug
 
 
 def get_type():
@@ -25,13 +24,14 @@ def get_compagny_name():
 def get_maintainer():
 	return ["Mr DUPIN Edouard <yui.heero@gmail.com>"]
 
-def create(target, module_name):
-	my_module = module.Module(__file__, module_name, get_type())
-	
+def configure(target, my_module):
 	my_module.add_src_file([
-		'orchestra-out.cpp'
-		])
-	my_module.add_module_depend(['audio-orchestra', 'test-debug'])
-	return my_module
+	    'orchestra-out.cpp'
+	    ])
+	my_module.add_depend([
+	    'audio-orchestra',
+	    'test-debug'
+	    ])
+	return True
 
 

tools/orchestra-in.cpp

@@ -4,20 +4,20 @@
  * @license APACHE v2.0 (see license file)
  */
 
-#include <etk/etk.h>
-#include <test-debug/debug.h>
-#include <unistd.h>
-#include <audio/orchestra/Interface.h>
+#include <etk/etk.hpp>
+#include <test-debug/debug.hpp>
+
+#include <audio/orchestra/Interface.hpp>
 
 int main(int _argc, const char **_argv) {
 	// the only one init for etk:
 	etk::init(_argc, _argv);
 	for (int32_t iii=0; iii<_argc ; ++iii) {
-		std::string data = _argv[iii];
+		etk::String data = _argv[iii];
 		if (    data == "-h"
 		     || data == "--help") {
-			std::cout << "Help : " << std::endl;
-			std::cout << "    ./xxx ---" << std::endl;
+			TEST_PRINT("Help : ");
+			TEST_PRINT("    ./xxx ---");
 			exit(0);
 		}
 	}

tools/orchestra-list.cpp

@@ -4,25 +4,25 @@
  * @license APACHE v2.0 (see license file)
  */
 
-#include <etk/etk.h>
-#include <test-debug/debug.h>
-#include <unistd.h>
-#include <audio/orchestra/Interface.h>
+#include <etk/etk.hpp>
+#include <test-debug/debug.hpp>
+
+#include <audio/orchestra/Interface.hpp>
 
 int main(int _argc, const char **_argv) {
 	// the only one init for etk:
 	etk::init(_argc, _argv);
 	for (int32_t iii=0; iii<_argc ; ++iii) {
-		std::string data = _argv[iii];
+		etk::String data = _argv[iii];
 		if (    data == "-h"
 		     || data == "--help") {
-			std::cout << "Help : " << std::endl;
-			std::cout << "    ./xxx ---" << std::endl;
+			TEST_PRINT("Help : ");
+			TEST_PRINT("    ./xxx ---");
 			exit(0);
 		}
 	}
 	audio::orchestra::Interface interface;
-	std::vector<std::string> apis = interface.getListApi();
+	etk::Vector<etk::String> apis = interface.getListApi();
 	TEST_PRINT("Find : " << apis.size() << " apis.");
 	for (auto &it : apis) {
 		interface.instanciate(it);

tools/orchestra-out.cpp

@@ -4,20 +4,20 @@
  * @license APACHE v2.0 (see license file)
  */
 
-#include <etk/etk.h>
-#include <test-debug/debug.h>
-#include <unistd.h>
-#include <audio/orchestra/Interface.h>
+#include <etk/etk.hpp>
+#include <test-debug/debug.hpp>
+
+#include <audio/orchestra/Interface.hpp>
 
 int main(int _argc, const char **_argv) {
 	// the only one init for etk:
 	etk::init(_argc, _argv);
 	for (int32_t iii=0; iii<_argc ; ++iii) {
-		std::string data = _argv[iii];
+		etk::String data = _argv[iii];
 		if (    data == "-h"
 		     || data == "--help") {
-			std::cout << "Help : " << std::endl;
-			std::cout << "    ./xxx ---" << std::endl;
+			TEST_PRINT("Help : ");
+			TEST_PRINT("    ./xxx ---");
 			exit(0);
 		}
 	}

version.txt

@@ -0,0 +1 @@
+1.0.0