.gitignore

@@ -1,64 +0,0 @@
-
-###################################
-# folders
-###################################
-CVS
-.svn
-Object_*
-doxygen/API/
-doxygen/ALL/
-
-###################################
-# backup files
-###################################
-*~
-*.swp
-*.old
-*.bck
-
-###################################
-# Compiled source #
-###################################
-*.com
-*.class
-*.dll
-*.exe
-*.o
-*.so
-*.pyc
-tags
-#ewol
-out
-ewol_debug
-ewol_release
-
-###################################
-# Packages #
-###################################
-# it's better to unpack these files and commit the raw source
-# git has its own built in compression methods
-*.7z
-*.dmg
-*.gz
-*.iso
-*.jar
-*.rar
-*.tar
-*.zip
-
-###################################
-# Logs and databases #
-###################################
-*.log
-*.sql
-*.sqlite
-
-###################################
-# OS generated files #
-###################################
-.DS_Store?
-ehthumbs.db
-Icon?
-Thumbs.db
-Sources/libewol/ewol/os/AndroidAbstraction.cpp
-org_ewol_EwolConstants.h

.travis.yml

@@ -1,7 +1,11 @@
-language: cpp
+language:
+  - cpp
 
-sudo: required
+sudo: false
-dist: trusty
+
+os:
+  - linux
+  - osx
 
 branches:
   only:
@@ -14,81 +18,33 @@ 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
-  - if [ "$TAG" == "Android" ]; then
+
-        git clone --depth 1 --branch master https://github.com/HeeroYui/android-download-tool;
+env:
-        ./android-download-tool/dl-android.sh;
+  - CONF=debug   BOARD=Linux BUILDER=clang GCOV=
-    fi
+  - CONF=release BOARD=Linux BUILDER=clang GCOV=
-  - git clone --depth 1 --branch master https://github.com/atria-soft/ci.git
+  - CONF=debug   BOARD=Linux BUILDER=gcc   GCOV=
-  - cd -
+  - CONF=release BOARD=Linux BUILDER=gcc   GCOV=
+  - CONF=debug   BOARD=Linux BUILDER=gcc   GCOV=--gcov
 
 before_script:
   - cd ..
-  - git clone https://github.com/atria-soft/etk.git -b $TRAVIS_BRANCH
+  - wget http://atria-soft.com/ci/coverage_send.py
-  - git clone https://github.com/atria-soft/elog.git -b $TRAVIS_BRANCH
+  - wget http://atria-soft.com/ci/test_send.py
-  - git clone https://github.com/atria-soft/ememory.git -b $TRAVIS_BRANCH
+  - wget http://atria-soft.com/ci/warning_send.py
-  - git clone https://github.com/atria-soft/echrono.git -b $TRAVIS_BRANCH
+  - git clone https://github.com/atria-soft/etk.git
-  - git clone https://github.com/atria-soft/ethread.git -b $TRAVIS_BRANCH
+  - git clone https://github.com/musicdsp/audio.git
-  - 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 $TRAVIS_BRANCH
-  - git clone https://github.com/generic-library/gtest-lutin.git --recursive
-  - git clone https://github.com/generic-library/z-lutin.git --recursive
   - pwd
   - ls -l
-  - if [ "$TRAVIS_OS_NAME" == "osx" ]; then
+  - if [ "$BUILDER" == "gcc" ]; then COMPILATOR_OPTION="--compilator-version=4.9"; else COMPILATOR_OPTION=""; fi
-        export PATH=$PATH:/Users/travis/Library/Python/2.7/bin/;
-    fi
-  - ./ci/build_send.py --tag=$TAG --status=START;
-
 
 script:
-  - lutin -w -j4 -C -P -t$TARGET -c $BUILDER $COMPILATOR_OPTION $BUS -m $CONF $GCOV $DISABLE_PACKAGE audio-orchestra; STATUS=$?
+  - lutin -w -j4 -C -P -c $BUILDER $COMPILATOR_OPTION -m $CONF $GCOV -p audio-orchestra
-  - ./ci/build_send.py --tag=$TAG --status="$STATUS";
+#  - ./out/Linux_x86_64/$CONF/staging/$BUILDER/audio-orchestra/usr/bin/audio-orchestra -l6
-
-after_script:
-  - if [ "$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
 
-
-

README.md

@@ -1,4 +1,4 @@
-# audio-orchestra
+------------------------------
-(MIT) audio: backend to acces hardware access (Fork of the original RTAudio lib)
+
+Nothing else ...
 
-[![Build Status](https://travis-ci.org/musicdsp/audio-orchestra.svg?branch=master)](https://travis-ci.org/musicdsp/audio-orchestra)

android/org/musicdsp/orchestra/OrchestraConstants.java

@@ -1,13 +0,0 @@
-/**
- * @author Edouard DUPIN
- *
- * @copyright 2015, Edouard DUPIN, all right reserved
- *
- * @license APACHE v2.0 (see license file)
- */
-
-package org.musicdsp.orchestra;
-
-public interface OrchestraConstants {
-	public static final int BUFFER_SIZE = 512;
-}

android/org/musicdsp/orchestra/OrchestraInterfaceInput.java

@@ -1,119 +0,0 @@
-/**
- * @author Edouard DUPIN, Kevin BILLONNEAU
- *
- * @copyright 2015, Edouard DUPIN, all right reserved
- *
- * @license APACHE v2.0 (see license file)
- */
-
-package org.musicdsp.orchestra;
-import android.media.AudioRecord;
-import android.media.AudioFormat;
-import android.media.AudioManager;
-import android.media.AudioRecord;
-import android.media.MediaRecorder;
-import android.util.Log;
-
-
-
-public class OrchestraInterfaceInput implements Runnable, OrchestraConstants {
-	private Thread m_thread = null;
-	private int m_uid = -1;
-	private OrchestraNative m_orchestraNativeHandle;
-	private boolean m_stop = false;
-	private boolean m_suspend = false;
-	private AudioRecord m_audio = null;
-	private int m_sampleRate = 48000;
-	private int m_nbChannel = 2;
-	private int m_format = 1;
-	private int m_bufferSize = BUFFER_SIZE;
-	
-	public OrchestraInterfaceInput(int _id, OrchestraNative _instance, int _idDevice, int _sampleRate, int _nbChannel, int _format) {
-		Log.d("InterfaceInput", "new: Input");
-		m_uid = _id;
-		m_orchestraNativeHandle = _instance;
-		m_stop = false;
-		m_suspend = false;
-		m_sampleRate = _sampleRate;
-		m_nbChannel = _nbChannel;
-		m_format = _format;
-		m_bufferSize = BUFFER_SIZE * m_nbChannel;
-	}
-	public int getUId() {
-		return m_uid;
-	}
-	
-	public void run() {
-		Log.e("InterfaceInput", "RUN (start)");
-		int channelConfig = AudioFormat.CHANNEL_CONFIGURATION_STEREO;
-		int audioFormat = AudioFormat.ENCODING_PCM_16BIT;
-		// we keep the minimum buffer size, otherwite the delay is too big ...
-		// TODO : int bufferSize = AudioRecord.getMinBufferSize(m_sampleRate, channelConfig, audioFormat);
-		int config = 0;
-		if (m_nbChannel == 1) {
-			config = AudioFormat.CHANNEL_IN_MONO;
-		} else {
-			config = AudioFormat.CHANNEL_IN_STEREO;
-		}
-		// Create a streaming AudioTrack for music playback
-		short[] streamBuffer = new short[m_bufferSize];
-		m_audio = new AudioRecord(MediaRecorder.AudioSource.MIC,
-		                          m_sampleRate,
-		                          config,
-		                          audioFormat,
-		                          m_bufferSize);
-		m_audio.startRecording();
-		
-		while (    m_stop == false
-		        && m_suspend == false) {
-			// Stream PCM data into the local buffer
-			m_audio.read(streamBuffer, 0, m_bufferSize);
-			// Send it to C++
-			m_orchestraNativeHandle.record(m_uid, streamBuffer, m_bufferSize/m_nbChannel);
-		}
-		m_audio.stop();
-		m_audio = null;
-		streamBuffer = null;
-		Log.e("InterfaceInput", "RUN (stop)");
-	}
-	
-	public void autoStart() {
-		m_stop=false;
-		if (m_suspend == false) {
-			Log.e("InterfaceInput", "Create thread");
-			m_thread = new Thread(this);
-			Log.e("InterfaceInput", "start thread");
-			m_thread.start();
-			Log.e("InterfaceInput", "start thread (done)");
-		}
-	}
-	
-	public void autoStop() {
-		if(m_audio == null) {
-			return;
-		}
-		m_stop=true;
-		m_thread = null;
-		/*
-		try {
-			super.join();
-		} catch(InterruptedException e) { }
-		*/
-	}
-	public void activityResume() {
-		m_suspend = false;
-		if (m_stop == false) {
-			Log.i("InterfaceInput", "Resume audio stream : " + m_uid);
-			m_thread = new Thread(this);
-			m_thread.start();
-		}
-	}
-	public void activityPause() {
-		if(m_audio == null) {
-			return;
-		}
-		m_suspend = true;
-		Log.i("InterfaceInput", "Pause audio stream : " + m_uid);
-		m_thread = null;
-	}
-}

android/org/musicdsp/orchestra/OrchestraInterfaceOutput.java

@@ -1,108 +0,0 @@
-/**
- * @author Edouard DUPIN, Kevin BILLONNEAU
- *
- * @copyright 2015, Edouard DUPIN, all right reserved
- *
- * @license APACHE v2.0 (see license file)
- */
-package org.musicdsp.orchestra;
-
-import android.media.AudioTrack;
-import android.media.AudioFormat;
-import android.media.AudioManager;
-import android.media.AudioRecord;
-import android.util.Log;
-
-
-public class OrchestraInterfaceOutput extends Thread implements OrchestraConstants {
-	private int m_uid = -1;
-	private OrchestraNative m_orchestraNativeHandle;
-	private boolean m_stop = false;
-	private boolean m_suspend = false;
-	private AudioTrack m_audio = null;
-	private int m_sampleRate = 48000;
-	private int m_nbChannel = 2;
-	private int m_format = 1;
-	private int m_bufferSize = BUFFER_SIZE;
-	
-	public OrchestraInterfaceOutput(int _id, OrchestraNative _instance, int _idDevice, int _sampleRate, int _nbChannel, int _format) {
-		Log.d("InterfaceOutput", "new: output");
-		m_uid = _id;
-		m_orchestraNativeHandle = _instance;
-		m_stop = true;
-		m_sampleRate = _sampleRate;
-		m_nbChannel = _nbChannel;
-		m_format = _format;
-		m_bufferSize = BUFFER_SIZE * m_nbChannel;
-	}
-	public int getUId() {
-		return m_uid;
-	}
-	
-	public void run() {
-		Log.e("InterfaceOutput", "RUN (start)");
-		int channelConfig = AudioFormat.CHANNEL_CONFIGURATION_STEREO;
-		int audioFormat = AudioFormat.ENCODING_PCM_16BIT;
-		// we keep the minimum buffer size, otherwite the delay is too big ...
-		//int bufferSize = AudioTrack.getMinBufferSize(m_sampleRate, channelConfig, audioFormat);
-		int config = 0;
-		if (m_nbChannel == 1) {
-			config = AudioFormat.CHANNEL_OUT_MONO;
-		} else if (m_nbChannel == 4) {
-			config = AudioFormat.CHANNEL_OUT_QUAD;
-		} else {
-			config = AudioFormat.CHANNEL_OUT_STEREO;
-		}
-		// Create a streaming AudioTrack for music playback
-		short[] streamBuffer = new short[m_bufferSize];
-		m_audio = new AudioTrack(AudioManager.STREAM_MUSIC,
-		                         m_sampleRate,
-		                         config,
-		                         audioFormat,
-		                         m_bufferSize,
-		                         AudioTrack.MODE_STREAM);
-		m_audio.play();
-		//m_audio.setPositionNotificationPeriod(2048);
-		
-		while (m_stop == false) {
-			// Fill buffer with PCM data from C++
-			m_orchestraNativeHandle.playback(m_uid, streamBuffer, m_bufferSize/m_nbChannel);
-			// Stream PCM data into the music AudioTrack
-			m_audio.write(streamBuffer, 0, m_bufferSize);
-		}
-		
-		m_audio.flush();
-		m_audio.stop();
-		m_audio = null;
-		streamBuffer = null;
-		Log.e("InterfaceOutput", "RUN (stop)");
-	}
-	public void autoStart() {
-		m_stop=false;
-		this.start();
-	}
-	public void autoStop() {
-		if(m_audio == null) {
-			return;
-		}
-		m_stop=true;
-		try {
-			super.join();
-		} catch(InterruptedException e) { }
-	}
-	public void activityResume() {
-		if (m_audio != null) {
-			Log.i("InterfaceOutput", "Resume audio stream : " + m_uid);
-			m_audio.play();
-		}
-	}
-	public void activityPause() {
-		if(m_audio == null) {
-			return;
-		}
-		if (m_audio != null) {
-			Log.i("InterfaceOutput", "Pause audio stream : " + m_uid);
-			m_audio.pause();
-		}
-	}
-}

android/org/musicdsp/orchestra/OrchestraManager.java

@@ -1,262 +0,0 @@
-/**
- * @author Edouard DUPIN
- *
- * @copyright 2015, Edouard DUPIN, all right reserved
- *
- * @license APACHE v2.0 (see license file)
- */
-
-package org.musicdsp.orchestra;
-
-import android.util.Log;
-import java.util.Vector;
-
-//import org.musicdsp.orchestra.Constants;
-//import org.musicdsp.orchestra.ManagerCallback;
-//import org.musicdsp.orchestra.Orchestra;
-//import org.musicdsp.orchestra.InterfaceOutput;
-//import org.musicdsp.orchestra.InterfaceInput;
-
-/**
- * @brief Class : 
- *
- */
-public class OrchestraManager implements OrchestraManagerCallback, OrchestraConstants {
-	private OrchestraNative m_orchestraHandle;
-	private int m_uid = 0;
-	private Vector<OrchestraInterfaceOutput> m_outputList;
-	private Vector<OrchestraInterfaceInput> m_inputList;
-	
-	public OrchestraManager() {
-		// set the java evironement in the C sources :
-		m_orchestraHandle = new OrchestraNative(this);
-		m_outputList = new Vector<OrchestraInterfaceOutput>();
-		m_inputList = new Vector<OrchestraInterfaceInput>();
-	}
-	
-	public int getDeviceCount() {
-		Log.e("Manager", "Get device List");
-		return 2;
-	}
-	
-	public String getDeviceProperty(int _idDevice) {
-		if (_idDevice == 0) {
-			return   "{\n"
-			       + "	name:'speaker',\n"
-			       + "	type:'output',\n"
-			       + "	sample-rate:[8000,16000,24000,32000,48000,96000],\n"
-			       + "	channels:['front-left','front-right'],\n"
-			       + "	format:['int16'],\n"
-			       + "	default:true\n"
-			       + "}";
-		} else if (_idDevice == 1) {
-			return   "{\n"
-			       + "	name:'microphone',\n"
-			       + "	type:'input',\n"
-			       + "	sample-rate:[8000,16000,24000,32000,48000,96000],\n"
-			       + "	channels:['front-left','front-right'],\n"
-			       + "	format:['int16'],\n"
-			       + "	default:true\n"
-			       + "}";
-		} else {
-			return "{}";
-		}
-	}
-	
-	public int openDeviceOutput(int _idDevice, int _freq, int _nbChannel, int _format) {
-		OrchestraInterfaceOutput iface = new OrchestraInterfaceOutput(m_uid, m_orchestraHandle, _idDevice, _freq, _nbChannel, _format);
-		m_uid++;
-		Log.e("Manager", "Open device Output: " + _idDevice + " with m_uid=" + (m_uid-1));
-		if (iface != null) {
-			m_outputList.add(iface);
-			Log.e("Manager", "Added element count=" + m_outputList.size());
-			return m_uid-1;
-		}
-		return -1;
-	}
-	
-	public int openDeviceInput(int _idDevice, int _freq, int _nbChannel, int _format) {
-		OrchestraInterfaceInput iface = new OrchestraInterfaceInput(m_uid, m_orchestraHandle, _idDevice, _freq, _nbChannel, _format);
-		m_uid++;
-		Log.e("Manager", "Open device Input: " + _idDevice + " with m_uid=" + (m_uid-1));
-		if (iface != null) {
-			m_inputList.add(iface);
-			return m_uid-1;
-		}
-		return -1;
-	}
-	
-	public boolean closeDevice(int _uniqueID) {
-		Log.e("Manager", "Close device : " + _uniqueID);
-		if (_uniqueID<0) {
-			Log.e("Manager", "Can not Close device with m_uid: " + _uniqueID);
-			return false;
-		}
-		// find the Element with his ID:
-		if (m_inputList != null) {
-			for (int iii=0; iii<m_inputList.size(); iii++) {
-				if (m_inputList.get(iii) == null) {
-					Log.e("Manager", "Null input element: " + iii);
-					continue;
-				}
-				if (m_inputList.get(iii).getUId() == _uniqueID) {
-					// find it ...
-					m_inputList.remove(iii);
-					return true;
-				}
-			}
-		}
-		if (m_outputList != null) {
-			for (int iii=0; iii<m_outputList.size(); iii++) {
-				if (m_outputList.get(iii) == null) {
-					Log.e("Manager", "Null input element: " + iii);
-					continue;
-				}
-				if (m_outputList.get(iii).getUId() == _uniqueID) {
-					// find it ...
-					m_outputList.remove(iii);
-					return true;
-				}
-			}
-		}
-		Log.e("Manager", "Can not start device with m_uid: " + _uniqueID + " Element does not exist ...");
-		return false;
-	}
-	
-	public boolean start(int _uniqueID) {
-		Log.e("Manager", "start device : " + _uniqueID);
-		if (_uniqueID<0) {
-			Log.e("Manager", "Can not start device with m_uid: " + _uniqueID);
-			return false;
-		}
-		// find the Element with his ID:
-		if (m_inputList != null) {
-			for (int iii=0; iii<m_inputList.size(); iii++) {
-				if (m_inputList.get(iii) == null) {
-					Log.e("Manager", "Null input element: " + iii);
-					continue;
-				}
-				if (m_inputList.get(iii).getUId() == _uniqueID) {
-					// find it ...
-					m_inputList.get(iii).autoStart();
-					return true;
-				}
-			}
-		}
-		if (m_outputList != null) {
-			for (int iii=0; iii<m_outputList.size(); iii++) {
-				if (m_outputList.get(iii) == null) {
-					Log.e("Manager", "Null input element: " + iii);
-					continue;
-				}
-				if (m_outputList.get(iii).getUId() == _uniqueID) {
-					// find it ...
-					m_outputList.get(iii).autoStart();
-					return true;
-				}
-			}
-		}
-		Log.e("Manager", "Can not start device with UID: " + _uniqueID + " Element does not exist ...");
-		return false;
-	}
-	
-	public boolean stop(int _uniqueID) {
-		Log.e("Manager", "stop device : " + _uniqueID);
-		if (_uniqueID<0) {
-			Log.e("Manager", "Can not stop device with UID: " + _uniqueID);
-			return false;
-		}
-		// find the Element with his ID:
-		if (m_inputList != null) {
-			for (int iii=0; iii<m_inputList.size(); iii++) {
-				if (m_inputList.get(iii) == null) {
-					Log.e("Manager", "Null input element: " + iii);
-					continue;
-				}
-				if (m_inputList.get(iii).getUId() == _uniqueID) {
-					// find it ...
-					m_inputList.get(iii).autoStop();
-					return true;
-				}
-			}
-		}
-		if (m_outputList != null) {
-			for (int iii=0; iii<m_outputList.size(); iii++) {
-				if (m_outputList.get(iii) == null) {
-					Log.e("Manager", "Null input element: " + iii);
-					continue;
-				}
-				if (m_outputList.get(iii).getUId() == _uniqueID) {
-					// find it ...
-					m_outputList.get(iii).autoStop();
-					return true;
-				}
-			}
-		}
-		Log.e("Manager", "Can not stop device with UID: " + _uniqueID + " Element does not exist ...");
-		return false;
-	}
-	public void onCreate() {
-		Log.w("Manager", "onCreate ...");
-		// nothing to do ...
-	}
-	public void onStart() {
-		Log.w("Manager", "onStart ...");
-		// nothing to do ...
-	}
-	public void onRestart() {
-		Log.w("Manager", "onRestart ...");
-		// nothing to do ...
-	}
-	public void onResume() {
-		Log.w("Manager", "onResume ...");
-		// find the Element with his ID:
-		if (m_inputList != null) {
-			for (int iii=0; iii<m_inputList.size(); iii++) {
-				if (m_inputList.get(iii) == null) {
-					Log.e("Manager", "Null input element: " + iii);
-					continue;
-				}
-				m_inputList.get(iii).activityResume();
-			}
-		}
-		if (m_outputList != null) {
-			for (int iii=0; iii<m_outputList.size(); iii++) {
-				if (m_outputList.get(iii) == null) {
-					Log.e("Manager", "Null input element: " + iii);
-					continue;
-				}
-				m_outputList.get(iii).activityResume();
-			}
-		}
-	}
-	public void onPause() {
-		Log.w("Manager", "onPause ...");
-		// find the Element with his ID:
-		if (m_inputList != null) {
-			for (int iii=0; iii<m_inputList.size(); iii++) {
-				if (m_inputList.get(iii) == null) {
-					Log.e("Manager", "Null input element: " + iii);
-					continue;
-				}
-				m_inputList.get(iii).activityPause();
-			}
-		}
-		if (m_outputList != null) {
-			for (int iii=0; iii<m_outputList.size(); iii++) {
-				if (m_outputList.get(iii) == null) {
-					Log.e("Manager", "Null input element: " + iii);
-					continue;
-				}
-				m_outputList.get(iii).activityPause();
-			}
-		}
-	}
-	public void onStop() {
-		Log.w("Manager", "onStop ...");
-	}
-	public void onDestroy() {
-		Log.w("Manager", "onDestroy ...");
-	}
-}
-

android/org/musicdsp/orchestra/OrchestraManagerCallback.java

@@ -1,19 +0,0 @@
-/**
- * @author Edouard DUPIN, Kevin BILLONNEAU
- *
- * @copyright 2015, Edouard DUPIN, all right reserved
- *
- * @license APACHE v2.0 (see license file)
- */
-
-package org.musicdsp.orchestra;
-
-public interface OrchestraManagerCallback {
-	public int getDeviceCount();
-	public String getDeviceProperty(int _idDevice);
-	public int openDeviceInput(int _idDevice, int _sampleRate, int _nbChannel, int _format);
-	public int openDeviceOutput(int _idDevice, int _sampleRate, int _nbChannel, int _format);
-	public boolean closeDevice(int _uniqueID);
-	public boolean start(int _uniqueID);
-	public boolean stop(int _uniqueID);
-}

android/org/musicdsp/orchestra/OrchestraNative.java

@@ -1,43 +0,0 @@
-/**
- * @author Edouard DUPIN, Kevin BILLONNEAU
- *
- * @copyright 2015, Edouard DUPIN, all right reserved
- *
- * @license APACHE v2.0 (see license file)
- */
-
-package org.musicdsp.orchestra;
-
-import java.lang.UnsatisfiedLinkError;
-import java.lang.RuntimeException;
-import android.util.Log;
-
-public class OrchestraNative {
-	public <T extends OrchestraManagerCallback> OrchestraNative(T _managerInstance) {
-		try {
-			NNsetJavaManager(_managerInstance);
-		} catch (java.lang.UnsatisfiedLinkError e) {
-			Log.e("Orchestra", "JNI binding not present ...");
-			throw new RuntimeException("Orchestra binding not present ...");
-		}
-		Log.d("Orchestra", "new ...");
-	}
-	
-	public void setManagerRemove() {
-		NNsetJavaManagerRemove();
-	}
-	
-	public void playback(int _flowId, short[] _bufferData, int _nbChunk) {
-		NNPlayback(_flowId, _bufferData, _nbChunk);
-	}
-	
-	public void record(int _flowId, short[] _bufferData, int _nbChunk) {
-		NNRecord(_flowId, _bufferData, _nbChunk);
-	}
-	
-	private native <T extends OrchestraManagerCallback> void NNsetJavaManager(T _managerInstance);
-	private native void NNsetJavaManagerRemove();
-	private native void NNPlayback(int _flowId, short[] _bufferData, int _nbChunk);
-	private native void NNRecord(int _flowId, short[] _bufferData, int _nbChunk);
-}
-

audio/orchestra/Api.cpp

@@ -1,478 +0,0 @@
-/** @file
- * @author Edouard DUPIN 
- * @copyright 2011, Edouard DUPIN, all right reserved
- * @license APACHE v2.0 (see license file)
- * @fork from RTAudio
- */
-
-//#include <etk/types.hpp>
-#include <audio/orchestra/Interface.hpp>
-#include <audio/orchestra/debug.hpp>
-#include <etk/types.hpp>
-
-// Static variable definitions.
-const etk::Vector<uint32_t>& audio::orchestra::genericSampleRate() {
-	static etk::Vector<uint32_t> list;
-	if (list.size() == 0) {
-		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(null),
-  m_deviceBuffer(null) {
-	m_device[0] = 11111;
-	m_device[1] = 11111;
-	m_state = audio::orchestra::state::closed;
-	m_mode = audio::orchestra::mode_unknow;
-}
-
-audio::orchestra::Api::~Api() {
-	
-}
-
-enum audio::orchestra::error audio::orchestra::Api::startStream() {
-	ATA_VERBOSE("Start Stream");
-	m_startTime = audio::Time::now();
-	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) {
-	if (m_state != audio::orchestra::state::closed) {
-		ATA_ERROR("a stream is already open!");
-		return audio::orchestra::error_invalidUse;
-	}
-	if (    _oParams != null
-	     && _oParams->nChannels < 1) {
-		ATA_ERROR("a non-null output StreamParameters structure cannot have an nChannels value less than one.");
-		return audio::orchestra::error_invalidUse;
-	}
-	if (    _iParams != null
-	     && _iParams->nChannels < 1) {
-		ATA_ERROR("a non-null input StreamParameters structure cannot have an nChannels value less than one.");
-		return audio::orchestra::error_invalidUse;
-	}
-	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) {
-		ATA_ERROR("'format' parameter value is undefined.");
-		return audio::orchestra::error_invalidUse;
-	}
-	uint32_t nDevices = getDeviceCount();
-	uint32_t oChannels = 0;
-	if (_oParams != null) {
-		oChannels = _oParams->nChannels;
-		if (    _oParams->deviceId >= nDevices
-		     && _oParams->deviceName == "") {
-			ATA_ERROR("output device parameter value is invalid.");
-			return audio::orchestra::error_invalidUse;
-		}
-	}
-	uint32_t iChannels = 0;
-	if (_iParams != null) {
-		iChannels = _iParams->nChannels;
-		if (    _iParams->deviceId >= nDevices
-		     && _iParams->deviceName == "") {
-			ATA_ERROR("input device parameter value is invalid.");
-			return audio::orchestra::error_invalidUse;
-		}
-	}
-	clearStreamInfo();
-	bool result;
-	if (oChannels > 0) {
-		if (_oParams->deviceId == -1) {
-			result = openName(_oParams->deviceName,
-			                  audio::orchestra::mode_output,
-			                  oChannels,
-			                  _oParams->firstChannel,
-			                  _sampleRate,
-			                  _format,
-			                  _bufferFrames,
-			                  _options);
-		} else {
-			result = open(_oParams->deviceId,
-			              audio::orchestra::mode_output,
-			              oChannels,
-			              _oParams->firstChannel,
-			              _sampleRate,
-			              _format,
-			              _bufferFrames,
-			              _options);
-		}
-		if (result == false) {
-			ATA_ERROR("system ERROR");
-			return audio::orchestra::error_systemError;
-		}
-	}
-	if (iChannels > 0) {
-		if (_iParams->deviceId == -1) {
-			result = openName(_iParams->deviceName,
-			                  audio::orchestra::mode_input,
-			                  iChannels,
-			                  _iParams->firstChannel,
-			                  _sampleRate,
-			                  _format,
-			                  _bufferFrames,
-			                  _options);
-		} else {
-			result = open(_iParams->deviceId,
-			              audio::orchestra::mode_input,
-			              iChannels,
-			              _iParams->firstChannel,
-			              _sampleRate,
-			              _format,
-			              _bufferFrames,
-			              _options);
-		}
-		if (result == false) {
-			if (oChannels > 0) {
-				closeStream();
-			}
-			ATA_ERROR("system error");
-			return audio::orchestra::error_systemError;
-		}
-	}
-	m_callback = _callback;
-	//_options.numberOfBuffers = m_nBuffers;
-	m_state = audio::orchestra::state::stopped;
-	return audio::orchestra::error_none;
-}
-
-uint32_t audio::orchestra::Api::getDefaultInputDevice() {
-	// Should be implemented in subclasses if possible.
-	return 0;
-}
-
-uint32_t audio::orchestra::Api::getDefaultOutputDevice() {
-	// Should be implemented in subclasses if possible.
-	return 0;
-}
-
-enum audio::orchestra::error audio::orchestra::Api::closeStream() {
-	ATA_VERBOSE("Close Stream");
-	// MUST be implemented in subclasses!
-	return audio::orchestra::error_none;
-}
-
-bool audio::orchestra::Api::open(uint32_t /*device*/,
-                                 audio::orchestra::mode /*mode*/,
-                                 uint32_t /*channels*/,
-                                 uint32_t /*firstChannel*/,
-                                 uint32_t /*sampleRate*/,
-                                 audio::format /*format*/,
-                                 uint32_t * /*bufferSize*/,
-                                 const audio::orchestra::StreamOptions& /*options*/) {
-	// MUST be implemented in subclasses!
-	return false;
-}
-
-void audio::orchestra::Api::tickStreamTime() {
-	//ATA_WARNING("tick : size=" << m_bufferSize << " rate=" << m_sampleRate << " time=" << audio::Duration((int64_t(m_bufferSize) * int64_t(1000000000)) / int64_t(m_sampleRate)).count());
-	//ATA_WARNING("  one element=" << audio::Duration((int64_t(1000000000)) / int64_t(m_sampleRate)).count());
-	m_duration += audio::Duration((int64_t(m_bufferSize) * int64_t(1000000000)) / int64_t(m_sampleRate));
-}
-
-long audio::orchestra::Api::getStreamLatency() {
-	if (verifyStream() != audio::orchestra::error_none) {
-		return 0;
-	}
-	long totalLatency = 0;
-	if (    m_mode == audio::orchestra::mode_output
-	     || m_mode == audio::orchestra::mode_duplex) {
-		totalLatency = m_latency[0];
-	}
-	if (    m_mode == audio::orchestra::mode_input
-	     || m_mode == audio::orchestra::mode_duplex) {
-		totalLatency += m_latency[1];
-	}
-	return totalLatency;
-}
-
-audio::Time audio::orchestra::Api::getStreamTime() {
-	if (verifyStream() != audio::orchestra::error_none) {
-		return audio::Time();
-	}
-	return m_startTime + m_duration;
-}
-
-uint32_t audio::orchestra::Api::getStreamSampleRate() {
-	if (verifyStream() != audio::orchestra::error_none) {
-		return 0;
-	}
-	return m_sampleRate;
-}
-
-enum audio::orchestra::error audio::orchestra::Api::verifyStream() {
-	if (m_state == audio::orchestra::state::closed) {
-		ATA_ERROR("a stream is not open!");
-		return audio::orchestra::error_invalidUse;
-	}
-	return audio::orchestra::error_none;
-}
-
-void audio::orchestra::Api::clearStreamInfo() {
-	m_mode = audio::orchestra::mode_unknow;
-	m_state = audio::orchestra::state::closed;
-	m_sampleRate = 0;
-	m_bufferSize = 0;
-	m_nBuffers = 0;
-	m_userFormat = audio::format_unknow;
-	m_startTime = audio::Time();
-	m_duration = audio::Duration(0);
-	m_deviceBuffer = null;
-	m_callback = null;
-	for (int32_t iii=0; iii<2; ++iii) {
-		m_device[iii] = 11111;
-		m_doConvertBuffer[iii] = false;
-		m_deviceInterleaved[iii] = true;
-		m_doByteSwap[iii] = false;
-		m_nUserChannels[iii] = 0;
-		m_nDeviceChannels[iii] = 0;
-		m_channelOffset[iii] = 0;
-		m_deviceFormat[iii] = audio::format_unknow;
-		m_latency[iii] = 0;
-		m_userBuffer[iii].clear();
-		m_convertInfo[iii].channels = 0;
-		m_convertInfo[iii].inJump = 0;
-		m_convertInfo[iii].outJump = 0;
-		m_convertInfo[iii].inFormat = audio::format_unknow;
-		m_convertInfo[iii].outFormat = audio::format_unknow;
-		m_convertInfo[iii].inOffset.clear();
-		m_convertInfo[iii].outOffset.clear();
-	}
-}
-
-void audio::orchestra::Api::setConvertInfo(audio::orchestra::mode _mode, uint32_t _firstChannel) {
-	int32_t idTable = audio::orchestra::modeToIdTable(_mode);
-	if (_mode == audio::orchestra::mode_input) { // convert device to user buffer
-		m_convertInfo[idTable].inJump = m_nDeviceChannels[1];
-		m_convertInfo[idTable].outJump = m_nUserChannels[1];
-		m_convertInfo[idTable].inFormat = m_deviceFormat[1];
-		m_convertInfo[idTable].outFormat = m_userFormat;
-	} else { // convert user to device buffer
-		m_convertInfo[idTable].inJump = m_nUserChannels[0];
-		m_convertInfo[idTable].outJump = m_nDeviceChannels[0];
-		m_convertInfo[idTable].inFormat = m_userFormat;
-		m_convertInfo[idTable].outFormat = m_deviceFormat[0];
-	}
-	if (m_convertInfo[idTable].inJump < m_convertInfo[idTable].outJump) {
-		m_convertInfo[idTable].channels = m_convertInfo[idTable].inJump;
-	} else {
-		m_convertInfo[idTable].channels = m_convertInfo[idTable].outJump;
-	}
-	// Set up the interleave/deinterleave offsets.
-	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.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.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.pushBack(kkk);
-			m_convertInfo[idTable].outOffset.pushBack(kkk);
-		}
-	}
-
-	// Add channel offset.
-	if (_firstChannel > 0) {
-		if (m_deviceInterleaved[idTable]) {
-			if (_mode == audio::orchestra::mode_output) {
-				for (int32_t kkk=0; kkk<m_convertInfo[idTable].channels; ++kkk) {
-					m_convertInfo[idTable].outOffset[kkk] += _firstChannel;
-				}
-			} else {
-				for (int32_t kkk=0; kkk<m_convertInfo[idTable].channels; ++kkk) {
-					m_convertInfo[idTable].inOffset[kkk] += _firstChannel;
-				}
-			}
-		} else {
-			if (_mode == audio::orchestra::mode_output) {
-				for (int32_t kkk=0; kkk<m_convertInfo[idTable].channels; ++kkk) {
-					m_convertInfo[idTable].outOffset[kkk] += (_firstChannel * m_bufferSize);
-				}
-			} else {
-				for (int32_t kkk=0; kkk<m_convertInfo[idTable].channels; ++kkk) {
-					m_convertInfo[idTable].inOffset[kkk] += (_firstChannel * m_bufferSize);
-				}
-			}
-		}
-	}
-}
-
-void audio::orchestra::Api::convertBuffer(char *_outBuffer, char *_inBuffer, audio::orchestra::ConvertInfo &_info) {
-	// This function does format conversion, input/output channel compensation, and
-	// data interleaving/deinterleaving.	24-bit integers are assumed to occupy
-	// the lower three bytes of a 32-bit integer.
-
-	// Clear our device buffer when in/out duplex device channels are different
-	if (    _outBuffer == m_deviceBuffer
-	     && m_mode == audio::orchestra::mode_duplex
-	     && m_nDeviceChannels[0] < m_nDeviceChannels[1]) {
-		memset(_outBuffer, 0, m_bufferSize * _info.outJump * audio::getFormatBytes(_info.outFormat));
-	}
-	switch (audio::getFormatBytes(_info.outFormat)) {
-		case 1:
-			{
-				uint8_t *out = reinterpret_cast<uint8_t*>(_outBuffer);
-				uint8_t *in = reinterpret_cast<uint8_t*>(_inBuffer);
-				for (size_t iii=0; iii<m_bufferSize; ++iii) {
-					for (size_t jjj=0; jjj<_info.channels; jjj++) {
-						out[_info.outOffset[jjj]] = in[_info.inOffset[jjj]];
-					}
-					in += _info.inJump;
-					out += _info.outJump;
-				}
-			}
-			break;
-		case 2:
-			{
-				uint16_t *out = reinterpret_cast<uint16_t*>(_outBuffer);
-				uint16_t *in = reinterpret_cast<uint16_t*>(_inBuffer);
-				for (size_t iii=0; iii<m_bufferSize; ++iii) {
-					for (size_t jjj=0; jjj<_info.channels; jjj++) {
-						out[_info.outOffset[jjj]] = in[_info.inOffset[jjj]];
-					}
-					in += _info.inJump;
-					out += _info.outJump;
-				}
-			}
-			break;
-		case 4:
-			{
-				uint32_t *out = reinterpret_cast<uint32_t*>(_outBuffer);
-				uint32_t *in = reinterpret_cast<uint32_t*>(_inBuffer);
-				for (size_t iii=0; iii<m_bufferSize; ++iii) {
-					for (size_t jjj=0; jjj<_info.channels; jjj++) {
-						out[_info.outOffset[jjj]] = in[_info.inOffset[jjj]];
-					}
-					in += _info.inJump;
-					out += _info.outJump;
-				}
-			}
-			break;
-		case 8:
-			{
-				uint64_t *out = reinterpret_cast<uint64_t*>(_outBuffer);
-				uint64_t *in = reinterpret_cast<uint64_t*>(_inBuffer);
-				for (size_t iii=0; iii<m_bufferSize; ++iii) {
-					for (size_t jjj=0; jjj<_info.channels; jjj++) {
-						out[_info.outOffset[jjj]] = in[_info.inOffset[jjj]];
-					}
-					in += _info.inJump;
-					out += _info.outJump;
-				}
-			}
-			break;
-	}
-}
-
-void audio::orchestra::Api::byteSwapBuffer(char *_buffer, uint32_t _samples, audio::format _format) {
-	char val;
-	char *ptr;
-	ptr = _buffer;
-	if (_format == audio::format_int16) {
-		for (uint32_t iii=0; iii<_samples; ++iii) {
-			// Swap 1st and 2nd bytes.
-			val = *(ptr);
-			*(ptr) = *(ptr+1);
-			*(ptr+1) = val;
-
-			// Increment 2 bytes.
-			ptr += 2;
-		}
-	} else if (    _format == audio::format_int32
-	            || _format == audio::format_float) {
-		for (uint32_t iii=0; iii<_samples; ++iii) {
-			// Swap 1st and 4th bytes.
-			val = *(ptr);
-			*(ptr) = *(ptr+3);
-			*(ptr+3) = val;
-
-			// Swap 2nd and 3rd bytes.
-			ptr += 1;
-			val = *(ptr);
-			*(ptr) = *(ptr+1);
-			*(ptr+1) = val;
-
-			// Increment 3 more bytes.
-			ptr += 3;
-		}
-	} else if (_format == audio::format_int24) {
-		for (uint32_t iii=0; iii<_samples; ++iii) {
-			// Swap 1st and 3rd bytes.
-			val = *(ptr);
-			*(ptr) = *(ptr+2);
-			*(ptr+2) = val;
-
-			// Increment 2 more bytes.
-			ptr += 2;
-		}
-	} else if (_format == audio::format_double) {
-		for (uint32_t iii=0; iii<_samples; ++iii) {
-			// Swap 1st and 8th bytes
-			val = *(ptr);
-			*(ptr) = *(ptr+7);
-			*(ptr+7) = val;
-
-			// Swap 2nd and 7th bytes
-			ptr += 1;
-			val = *(ptr);
-			*(ptr) = *(ptr+5);
-			*(ptr+5) = val;
-
-			// Swap 3rd and 6th bytes
-			ptr += 1;
-			val = *(ptr);
-			*(ptr) = *(ptr+3);
-			*(ptr+3) = val;
-
-			// Swap 4th and 5th bytes
-			ptr += 1;
-			val = *(ptr);
-			*(ptr) = *(ptr+1);
-			*(ptr+1) = val;
-
-			// Increment 5 more bytes.
-			ptr += 5;
-		}
-	}
-}
-
-

audio/orchestra/Api.hpp

@@ -1,180 +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/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 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 (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 (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 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:
-				int32_t channels;
-				int32_t inJump;
-				int32_t outJump;
-				enum audio::format inFormat;
-				enum audio::format outFormat;
-				etk::Vector<int> inOffset;
-				etk::Vector<int> outOffset;
-		};
-	
-		class Api : public ememory::EnableSharedFromThis<Api>{
-			protected:
-				etk::String m_name;
-			public:
-				Api();
-				virtual ~Api();
-				void setName(const etk::String& _name) {
-					m_name = _name;
-				}
-				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 etk::String& _deviceName, audio::orchestra::DeviceInfo& _info) {
-					return false;
-				}
-				virtual uint32_t getDefaultInputDevice();
-				virtual uint32_t getDefaultOutputDevice();
-				enum audio::orchestra::error openStream(audio::orchestra::StreamParameters* _outputParameters,
-				                                        audio::orchestra::StreamParameters* _inputParameters,
-				                                        audio::format _format,
-				                                        uint32_t _sampleRate,
-				                                        uint32_t* _nbChunk,
-				                                        audio::orchestra::AirTAudioCallback _callback,
-				                                        const audio::orchestra::StreamOptions& _options);
-				virtual enum audio::orchestra::error closeStream();
-				virtual enum audio::orchestra::error startStream();
-				virtual enum audio::orchestra::error stopStream() = 0;
-				virtual enum audio::orchestra::error abortStream() = 0;
-				long getStreamLatency();
-				uint32_t getStreamSampleRate();
-				virtual audio::Time getStreamTime();
-				bool isStreamOpen() const {
-					return m_state != audio::orchestra::state::closed;
-				}
-				bool isStreamRunning() const {
-					return m_state == audio::orchestra::state::running;
-				}
-				
-			protected:
-				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
-				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.
-				bool m_doByteSwap[2]; // Playback and record, respectively.
-				uint32_t m_sampleRate; // TODO : Rename frequency
-				uint32_t m_bufferSize;
-				uint32_t m_nBuffers;
-				uint32_t m_nUserChannels[2]; // Playback and record, respectively. // TODO : set only one config (open inout with the same number of channels (limitation)
-				uint32_t m_nDeviceChannels[2]; // Playback and record channels, respectively.
-				uint32_t m_channelOffset[2]; // Playback and record, respectively.
-				uint64_t m_latency[2]; // Playback and record, respectively.
-				enum audio::format m_userFormat; // TODO : Remove this ==> use can only open in the Harware format ...
-				enum audio::format m_deviceFormat[2]; // Playback and record, respectively.
-				audio::orchestra::ConvertInfo m_convertInfo[2];
-				
-				//audio::Time
-				audio::Time m_startTime; //!< start time of the stream (restart at every stop, pause ...)
-				audio::Duration m_duration; //!< duration from wich the stream is started
-				
-				/**
-				 * @brief api-specific method that attempts to open a device
-				 * with the given parameters. This function MUST be implemented by
-				 * all subclasses. If an error is encountered during the probe, a
-				 * "warning" message is reported and false is returned. A
-				 * successful probe is indicated by a return value of true.
-				 */
-				virtual bool open(uint32_t _device,
-				                  enum audio::orchestra::mode _mode,
-				                  uint32_t _channels,
-				                  uint32_t _firstChannel,
-				                  uint32_t _sampleRate,
-				                  enum audio::format _format,
-				                  uint32_t *_bufferSize,
-				                  const audio::orchestra::StreamOptions& _options);
-				virtual bool openName(const etk::String& _deviceName,
-				                      audio::orchestra::mode _mode,
-				                      uint32_t _channels,
-				                      uint32_t _firstChannel,
-				                      uint32_t _sampleRate,
-				                      audio::format _format,
-				                      uint32_t *_bufferSize,
-				                                 const audio::orchestra::StreamOptions& _options) { return false; }
-				/**
-				 * @brief Increment the stream time.
-				 */
-				void tickStreamTime();
-				/**
-				 * @brief Clear an RtApiStream structure.
-				 */
-				void clearStreamInfo();
-				/**
-				 * @brief Check the current stream status
-				 */
-				enum audio::orchestra::error verifyStream();
-				/**
-				 * @brief Protected method used to perform format, channel number, and/or interleaving
-				 * conversions between the user and device buffers.
-				 */
-				void convertBuffer(char *_outBuffer,
-				                   char *_inBuffer,
-				                   audio::orchestra::ConvertInfo& _info);
-				
-				/**
-				 * @brief Perform byte-swapping on buffers.
-				 */
-				void byteSwapBuffer(char *_buffer,
-				                    uint32_t _samples,
-				                    enum audio::format _format);
-				/**
-				 * @brief Sets up the parameters for buffer conversion.
-				 */
-				void setConvertInfo(enum audio::orchestra::mode _mode,
-				                    uint32_t _firstChannel);
-				
-			public:
-				virtual bool isMasterOf(ememory::SharedPtr<audio::orchestra::Api> _api) {
-					return false;
-				};
-		};
-	}
-}
-

audio/orchestra/DeviceInfo.cpp

@@ -1,61 +0,0 @@
-/** @file
- * @author Edouard DUPIN 
- * @copyright 2011, Edouard DUPIN, all right reserved
- * @license APACHE v2.0 (see license file)
- * @fork from RTAudio
- */
-
-//#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 {
-	etk::String space;
-	for (int32_t iii=0; iii<_tabNumber; ++iii) {
-		space += "    ";
-	}
-	if (isCorrect == false) {
-		ATA_PRINT(space + "NOT CORRECT INFORAMATIONS");
-		return;
-	}
-	ATA_PRINT(space + "mode=" << (input==true?"input":"output"));
-	ATA_PRINT(space + "name=" << name);
-	if (desc.size() != 0) {
-		ATA_PRINT(space + "desc=" << desc);
-	}
-	ATA_PRINT(space + "channel" << (channels.size()>1?"s":"") << "=" << channels.size() << " : " << channels);
-	ATA_PRINT(space + "rate" << (sampleRates.size()>1?"s":"") << "=" << sampleRates);
-	ATA_PRINT(space + "native Format" << (nativeFormats.size()>1?"s":"") << ": " << nativeFormats);
-	ATA_PRINT(space + "default=" << (isDefault==true?"true":"false"));
-}
-
-void audio::orchestra::DeviceInfo::clear() {
-	isCorrect = false;
-	input = false;
-	name = "";
-	desc = "";
-	channels.clear();
-	sampleRates.clear();
-	nativeFormats.clear();
-	isDefault = false;
-}
-
-etk::Stream& audio::orchestra::operator <<(etk::Stream& _os, const audio::orchestra::DeviceInfo& _obj) {
-	_os << "{";
-	if (_obj.isCorrect == false) {
-		_os << "NOT CORRECT INFORAMATIONS";
-	} else {
-		_os << "name=" << _obj.name << ", ";
-		if (_obj.desc.size() != 0) {
-			_os << "description=" << _obj.desc << ", ";
-		}
-		_os << "channels=" << _obj.channels << ", ";
-		_os << "default=" << _obj.isDefault << ", ";
-		_os << "rates=" << _obj.sampleRates << ", ";
-		_os << "native Format: " << _obj.nativeFormats;
-	}
-	_os << "}";
-	return _os;
-}
-

audio/orchestra/DeviceInfo.hpp

@@ -1,49 +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 <audio/format.hpp>
-#include <audio/channel.hpp>
-
-namespace audio {
-	namespace orchestra {
-		/**
-		 * @brief The public device information structure for returning queried values.
-		 */
-		class DeviceInfo {
-			public:
-				bool isCorrect; //!< the information is correct (the system can return information incorect).
-				bool input; //!< true if the device in an input; false: output.
-				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() :
-				  isCorrect(false),
-				  input(false),
-				  name(),
-				  desc(),
-				  channels(),
-				  sampleRates(),
-				  nativeFormats(),
-				  isDefault(false) {}
-				/**
-				 * @brief Display the current information of the device (on console)
-				 */
-				void display(int32_t _tabNumber = 1) const;
-				/**
-				 * @brief Clear all internal data
-				 */
-				void clear();
-		};
-		etk::Stream& operator <<(etk::Stream& _os, const audio::orchestra::DeviceInfo& _obj);
-	}
-}
-

audio/orchestra/Flags.cpp

@@ -1,9 +0,0 @@
-/** @file
- * @author Edouard DUPIN 
- * @copyright 2011, Edouard DUPIN, all right reserved
- * @license APACHE v2.0 (see license file)
- * @fork from RTAudio
- */
-
-#include <audio/orchestra/Flags.hpp>
-#include <audio/orchestra/debug.hpp>

audio/orchestra/Flags.hpp

@@ -1,22 +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.hpp>
-
-namespace audio {
-	namespace orchestra {
-		class Flags {
-			public:
-				bool m_minimizeLatency; // Simple example ==> TODO ...
-				Flags() :
-				  m_minimizeLatency(false) {
-					// nothing to do ...
-				}
-		};
-	}
-}

audio/orchestra/Interface.cpp

@@ -1,186 +0,0 @@
-/** @file
- * @author Edouard DUPIN 
- * @copyright 2011, Edouard DUPIN, all right reserved
- * @license APACHE v2.0 (see license file)
- * @fork from RTAudio
- */
-
-//#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>
-
-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.pushBack(m_apiAvaillable[iii].first);
-	}
-	return apis;
-}
-
-
-
-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 != null) {
-				return;
-			}
-		}
-	}
-	// TODO : An error occured ...
-	ATA_ERROR("Error in open API ...");
-}
-
-
-audio::orchestra::Interface::Interface() :
-  m_api(null) {
-	ATA_DEBUG("Add interface:");
-#if defined(ORCHESTRA_BUILD_JACK)
-	addInterface(audio::orchestra::typeJack, audio::orchestra::api::Jack::create);
-#endif
-#if defined(ORCHESTRA_BUILD_ALSA)
-	addInterface(audio::orchestra::typeAlsa, audio::orchestra::api::Alsa::create);
-#endif
-#if defined(ORCHESTRA_BUILD_PULSE)
-	addInterface(audio::orchestra::typePulse, audio::orchestra::api::Pulse::create);
-#endif
-#if defined(ORCHESTRA_BUILD_ASIO)
-	addInterface(audio::orchestra::typeAsio, audio::orchestra::api::Asio::create);
-#endif
-#if defined(ORCHESTRA_BUILD_DS)
-	addInterface(audio::orchestra::typeDs, audio::orchestra::api::Ds::create);
-#endif
-#if defined(ORCHESTRA_BUILD_MACOSX_CORE)
-	addInterface(audio::orchestra::typeCoreOSX, audio::orchestra::api::Core::create);
-#endif
-#if defined(ORCHESTRA_BUILD_IOS_CORE)
-	addInterface(audio::orchestra::typeCoreIOS, audio::orchestra::api::CoreIos::create);
-#endif
-#if defined(ORCHESTRA_BUILD_JAVA)
-	addInterface(audio::orchestra::typeJava, audio::orchestra::api::Android::create);
-#endif
-#if defined(ORCHESTRA_BUILD_DUMMY)
-	addInterface(audio::orchestra::typeDummy, audio::orchestra::api::Dummy::create);
-#endif
-}
-
-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 == null) {
-		ATA_WARNING("Interface NOT started!");
-		return audio::orchestra::error_none;
-	}
-	m_api.reset();
-	return audio::orchestra::error_none;
-}
-
-enum audio::orchestra::error audio::orchestra::Interface::instanciate(const etk::String& _api) {
-	ATA_INFO("Instanciate API ...");
-	if (m_api != null) {
-		ATA_WARNING("Interface already started!");
-		return audio::orchestra::error_none;
-	}
-	if (_api != audio::orchestra::typeUndefined) {
-		ATA_INFO("API specified : " << _api);
-		// Attempt to open the specified API.
-		openApi(_api);
-		if (m_api != null) {
-			if (m_api->getDeviceCount() != 0) {
-				ATA_INFO("    ==> api open");
-			}
-			return audio::orchestra::error_none;
-		}
-		// No compiled support for specified API value.	Issue a debug
-		// warning and continue as if no API was specified.
-		ATA_ERROR("API NOT Supported '" << _api << "' not in " << getListApi());
-		return audio::orchestra::error_fail;
-	}
-	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.
-	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 == null) {
-			ATA_ERROR("    ==> can not create ...");
-			continue;
-		}
-		if (m_api->getDeviceCount() != 0) {
-			ATA_INFO("    ==> api open");
-			break;
-		} else {
-			ATA_INFO("    ==> Interface exist, but have no devices: " << m_api->getDeviceCount());
-		}
-	}
-	if (m_api != null) {
-		return audio::orchestra::error_none;
-	}
-	ATA_ERROR("API NOT Supported '" << _api << "' not in " << getListApi());
-	return audio::orchestra::error_fail;
-}
-
-audio::orchestra::Interface::~Interface() {
-	ATA_INFO("Remove interface");
-	m_api.reset();
-}
-
-enum audio::orchestra::error audio::orchestra::Interface::openStream(audio::orchestra::StreamParameters* _outputParameters,
-                                                                     audio::orchestra::StreamParameters* _inputParameters,
-                                                                     audio::format _format,
-                                                                     uint32_t _sampleRate,
-                                                                     uint32_t* _bufferFrames,
-                                                                     audio::orchestra::AirTAudioCallback _callback,
-                                                                     const audio::orchestra::StreamOptions& _options) {
-	if (m_api == null) {
-		return audio::orchestra::error_inputNull;
-	}
-	return m_api->openStream(_outputParameters,
-	                           _inputParameters,
-	                           _format,
-	                           _sampleRate,
-	                           _bufferFrames,
-	                           _callback,
-	                           _options);
-}
-
-bool audio::orchestra::Interface::isMasterOf(audio::orchestra::Interface& _interface) {
-	if (m_api == null) {
-		ATA_ERROR("Current Master API is null ...");
-		return false;
-	}
-	if (_interface.m_api == null) {
-		ATA_ERROR("Current Slave API is null ...");
-		return false;
-	}
-	if (m_api->getCurrentApi() != _interface.m_api->getCurrentApi()) {
-		ATA_ERROR("Can not link 2 Interface with not the same Low level type (?)");//" << _interface.m_adac->getCurrentApi() << " != " << m_adac->getCurrentApi() << ")");
-		return false;
-	}
-	if (m_api->getCurrentApi() != audio::orchestra::typeAlsa) {
-		ATA_ERROR("Link 2 device together work only if the interafec is ?");// << audio::orchestra::type::alsa << " not for " << m_api->getCurrentApi());
-		return false;
-	}
-	return m_api->isMasterOf(_interface.m_api);
-}
-

audio/orchestra/Interface.hpp

@@ -1,308 +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/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 {
-		/**
-		 * @brief audio::orchestra::Interface class declaration.
-		 *
-		 * audio::orchestra::Interface is a "controller" used to select an available audio i/o
-		 * interface.	It presents a common API for the user to call but all
-		 * functionality is implemented by the class RtApi and its
-		 * subclasses.	RtAudio creates an instance of an RtApi subclass
-		 * based on the user's API choice.	If no choice is made, RtAudio
-		 * attempts to make a "logical" API selection.
-		 */
-		class Interface {
-			protected:
-				etk::Vector<etk::Pair<etk::String, ememory::SharedPtr<Api> (*)()> > m_apiAvaillable;
-			protected:
-				ememory::SharedPtr<audio::orchestra::Api> m_api;
-			public:
-				void setName(const etk::String& _name) {
-					if (m_api == null) {
-						return;
-					}
-					m_api->setName(_name);
-				}
-				/**
-				 * @brief Get the list of all availlable API in the system.
-				 * @return the list of all APIs
-				 */
-				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 etk::String& _api, ememory::SharedPtr<Api> (*_callbackCreate)());
-				/**
-				 * @brief The class constructor.
-				 * @note the creating of the basic instance is done by Instanciate
-				 */
-				Interface();
-				/**
-				 * @brief The destructor.
-				 * 
-				 * If a stream is running or open, it will be stopped and closed
-				 * automatically.
-				 */
-				virtual ~Interface();
-				/**
-				 * @brief Clear the current Interface
-				 */
-				enum audio::orchestra::error clear();
-				/**
-				 * @brief Create an interface instance
-				 */
-				enum audio::orchestra::error instanciate(const etk::String& _api = audio::orchestra::typeUndefined);
-				/**
-				 * @return the audio API specifier for the current instance of airtaudio.
-				 */
-				const etk::String& getCurrentApi() {
-					if (m_api == null) {
-						return audio::orchestra::typeUndefined;
-					}
-					return m_api->getCurrentApi();
-				}
-				/**
-				 * @brief A public function that queries for the number of audio devices available.
-				 * 
-				 * This function performs a system query of available devices each time it
-				 * is called, thus supporting devices connected \e after instantiation. If
-				 * a system error occurs during processing, a warning will be issued. 
-				 */
-				uint32_t getDeviceCount() {
-					if (m_api == null) {
-						return 0;
-					}
-					return m_api->getDeviceCount();
-				}
-				/**
-				 * @brief Any device integer between 0 and getDeviceCount() - 1 is valid.
-				 * If an invalid argument is provided, an RtError (type = INVALID_USE)
-				 * will be thrown.	If a device is busy or otherwise unavailable, the
-				 * structure member "probed" will have a value of "false" and all
-				 * other members are undefined.	If the specified device is the
-				 * current default input or output device, the corresponding
-				 * "isDefault" member will have a value of "true".
-				 *
-				 * @return An audio::orchestra::DeviceInfo structure for a specified device number.
-				 */
-				audio::orchestra::DeviceInfo getDeviceInfo(uint32_t _device) {
-					if (m_api == null) {
-						return audio::orchestra::DeviceInfo();
-					}
-					return m_api->getDeviceInfo(_device);
-				}
-				audio::orchestra::DeviceInfo getDeviceInfo(const etk::String& _deviceName) {
-					if (m_api == null) {
-						return audio::orchestra::DeviceInfo();
-					}
-					audio::orchestra::DeviceInfo info;
-					m_api->getNamedDeviceInfo(_deviceName, info);
-					return info;
-				}
-				/**
-				 * @brief A function that returns the index of the default output device.
-				 * 
-				 * If the underlying audio API does not provide a "default
-				 * device", or if no devices are available, the return value will be
-				 * 0. Note that this is a valid device identifier and it is the
-				 * client's responsibility to verify that a device is available
-				 * before attempting to open a stream.
-				 */
-				uint32_t getDefaultOutputDevice() {
-					if (m_api == null) {
-						return 0;
-					}
-					return m_api->getDefaultOutputDevice();
-				}
-				/**
-				 * @brief A function that returns the index of the default input device.
-				 * 
-				 * If the underlying audio API does not provide a "default
-				 * device", or if no devices are available, the return value will be
-				 * 0. Note that this is a valid device identifier and it is the
-				 * client's responsibility to verify that a device is available
-				 * before attempting to open a stream.
-				 */
-				uint32_t getDefaultInputDevice() {
-					if (m_api == null) {
-						return 0;
-					}
-					return m_api->getDefaultInputDevice();
-				}
-				/**
-				 * @brief A public function for opening a stream with the specified parameters.
-				 * 
-				 * An RtError (type = SYSTEM_ERROR) is thrown if a stream cannot be
-				 * opened with the specified parameters or an error occurs during
-				 * processing.	An RtError (type = INVALID_USE) is thrown if any
-				 * invalid device ID or channel number parameters are specified.
-				 * @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 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 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).
-				 * @param  _bufferFrames A pointer to a value indicating the desired
-				 *           internal buffer size in sample frames.	The actual value
-				 *           used by the device is returned via the same pointer. A
-				 *           value of zero can be specified, in which case the lowest
-				 *           allowable value is determined.
-				 * @param  _callback A client-defined function that will be invoked
-				 *           when input data is available and/or output data is needed.
-				 * @param  _options An optional pointer to a structure containing various
-				 *           global stream options, including a list of OR'ed audio::orchestra::streamFlags
-				 *           and a suggested number of stream buffers that can be used to 
-				 *           control stream latency. More buffers typically result in more
-				 *           robust performance, though at a cost of greater latency. If a
-				 *           value of zero is specified, a system-specific median value is
-				 *           chosen. If the airtaudio_MINIMIZE_LATENCY flag bit is set, the
-				 *           lowest allowable value is used. The actual value used is
-				 *           returned via the structure argument. The parameter is API dependent.
-				 * @param  _errorCallback A client-defined function that will be invoked
-				 *           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());
-				
-				/**
-				 * @brief A function that closes a stream and frees any associated stream memory.
-				 * 
-				 * If a stream is not open, this function issues a warning and
-				 * returns (no exception is thrown).
-				 */
-				enum audio::orchestra::error closeStream() {
-					if (m_api == null) {
-						return audio::orchestra::error_inputNull;
-					}
-					return m_api->closeStream();
-				}
-				/**
-				 * @brief A function that starts a stream.
-				 *
-				 * An RtError (type = SYSTEM_ERROR) is thrown if an error occurs
-				 * during processing.	An RtError (type = INVALID_USE) is thrown if a
-				 * stream is not open.	A warning is issued if the stream is already
-				 * running.
-				 */
-				enum audio::orchestra::error startStream() {
-					if (m_api == null) {
-						return audio::orchestra::error_inputNull;
-					}
-					return m_api->startStream();
-				}
-				/**
-				 * @brief Stop a stream, allowing any samples remaining in the output queue to be played.
-				 * 
-				 * An RtError (type = SYSTEM_ERROR) is thrown if an error occurs
-				 * during processing.	An RtError (type = INVALID_USE) is thrown if a
-				 * stream is not open.	A warning is issued if the stream is already
-				 * stopped.
-				*/
-				enum audio::orchestra::error stopStream() {
-					if (m_api == null) {
-						return audio::orchestra::error_inputNull;
-					}
-					return m_api->stopStream();
-				}
-				/**
-				 * @brief Stop a stream, discarding any samples remaining in the input/output queue.
-				 * An RtError (type = SYSTEM_ERROR) is thrown if an error occurs
-				 * during processing.	An RtError (type = INVALID_USE) is thrown if a
-				 * stream is not open.	A warning is issued if the stream is already
-				 * stopped.
-				 */
-				enum audio::orchestra::error abortStream() {
-					if (m_api == null) {
-						return audio::orchestra::error_inputNull;
-					}
-					return m_api->abortStream();
-				}
-				/**
-				 * @return true if a stream is open and false if not.
-				 */
-				bool isStreamOpen() const {
-					if (m_api == null) {
-						return false;
-					}
-					return m_api->isStreamOpen();
-				}
-				/**
-				 * @return true if the stream is running and false if it is stopped or not open.
-				 */
-				bool isStreamRunning() const {
-					if (m_api == null) {
-						return false;
-					}
-					return m_api->isStreamRunning();
-				}
-				/**
-				 * @brief If a stream is not open, an RtError (type = INVALID_USE) will be thrown.
-				 * @return the number of elapsed seconds since the stream was started.
-				 */
-				audio::Time getStreamTime() {
-					if (m_api == null) {
-						return audio::Time();
-					}
-					return m_api->getStreamTime();
-				}
-				/**
-				 * @brief The stream latency refers to delay in audio input and/or output
-				 * caused by internal buffering by the audio system and/or hardware.
-				 * For duplex streams, the returned value will represent the sum of
-				 * the input and output latencies.	If a stream is not open, an
-				 * RtError (type = INVALID_USE) will be thrown.	If the API does not
-				 * report latency, the return value will be zero.
-				 * @return The internal stream latency in sample frames.
-				 */
-				long getStreamLatency() {
-					if (m_api == null) {
-						return 0;
-					}
-					return m_api->getStreamLatency();
-				}
-				/**
-				 * @brief On some systems, the sample rate used may be slightly different
-				 * than that specified in the stream parameters. If a stream is not
-				 * open, an RtError (type = INVALID_USE) will be thrown.
-				 * @return Returns actual sample rate in use by the stream.
-				 */
-				uint32_t getStreamSampleRate() {
-					if (m_api == null) {
-						return 0;
-					}
-					return m_api->getStreamSampleRate();
-				}
-				bool isMasterOf(audio::orchestra::Interface& _interface);
-			protected:
-				void openApi(const etk::String& _api);
-		};
-	}
-}
-

audio/orchestra/StreamOptions.cpp

@@ -1,45 +0,0 @@
-/** @file
- * @author Edouard DUPIN 
- * @copyright 2011, Edouard DUPIN, all right reserved
- * @license APACHE v2.0 (see license file)
- * @fork from RTAudio
- */
-
-#include <audio/orchestra/StreamOptions.hpp>
-#include <etk/stdTools.hpp>
-#include <audio/orchestra/debug.hpp>
-
-static const char* listValue[] = {
-	"hardware",
-	"trigered",
-	"soft"
-};
-
-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 etk::String& _value) {
-		if (_value == "hardware") {
-			_variableRet = audio::orchestra::timestampMode_Hardware;
-			return true;
-		}
-		if (_value == "trigered") {
-			_variableRet = audio::orchestra::timestampMode_trigered;
-			return true;
-		}
-		if (_value == "soft") {
-			_variableRet = audio::orchestra::timestampMode_soft;
-			return true;
-		}
-		return false;
-	}
-	
-	template <enum audio::orchestra::timestampMode> etk::String toString(const enum audio::orchestra::timestampMode& _variable) {
-		return listValue[_variable];
-	}
-}
-
-

audio/orchestra/StreamOptions.hpp

@@ -1,35 +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 <audio/orchestra/Flags.hpp>
-#include <etk/String.hpp>
-
-namespace audio {
-	namespace orchestra {
-		enum timestampMode {
-			timestampMode_Hardware, //!< enable harware timestamp
-			timestampMode_trigered, //!< get harware triger time stamp and increment with duration
-			timestampMode_soft, //!< Simulate all timestamp.
-		};
-		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.
-				etk::String streamName; //!< A stream name (currently used only in Jack).
-				enum timestampMode mode; //!< mode of timestamping data...
-				// Default constructor.
-				StreamOptions() :
-				  flags(),
-				  numberOfBuffers(0),
-				  mode(timestampMode_Hardware) {}
-		};
-	}
-}
-

audio/orchestra/StreamParameters.hpp

@@ -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
-
-namespace audio {
-	namespace orchestra {
-		/**
-		 * @brief The structure for specifying input or ouput stream parameters.
-		 */
-		class StreamParameters {
-			public:
-				int32_t deviceId; //!< Device index (-1 to getDeviceCount() - 1).
-				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.
-				StreamParameters() :
-				  deviceId(-1),
-				  nChannels(0),
-				  firstChannel(0) {
-					
-				}
-		};
-	}
-}
-

audio/orchestra/api/Alsa.hpp

@@ -1,79 +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
-
-#ifdef ORCHESTRA_BUILD_ALSA
-
-namespace audio {
-	namespace orchestra {
-		namespace api {
-			class AlsaPrivate;
-			class Alsa: public audio::orchestra::Api {
-				public:
-					static ememory::SharedPtr<audio::orchestra::Api> create();
-				public:
-					Alsa();
-					virtual ~Alsa();
-					const etk::String& getCurrentApi() {
-						return audio::orchestra::typeAlsa;
-					}
-					uint32_t getDeviceCount();
-				private:
-					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 etk::String& _deviceName, audio::orchestra::DeviceInfo& _info) {
-						return getNamedDeviceInfoLocal(_deviceName, _info);
-					}
-					audio::orchestra::DeviceInfo getDeviceInfo(uint32_t _device);
-					enum audio::orchestra::error closeStream();
-					enum audio::orchestra::error startStream();
-					enum audio::orchestra::error stopStream();
-					enum audio::orchestra::error abortStream();
-					// This function is intended for internal use only.	It must be
-					// public because it is called by the internal callback handler,
-					// which is not a member of RtAudio.	External use of this function
-					// will most likely produce highly undesireable results!
-					void callbackEvent();
-					void callbackEventOneCycleRead();
-					void callbackEventOneCycleWrite();
-					void callbackEventOneCycleMMAPRead();
-					void callbackEventOneCycleMMAPWrite();
-				private:
-					ememory::SharedPtr<AlsaPrivate> m_private;
-					etk::Vector<audio::orchestra::DeviceInfo> m_devices;
-					void saveDeviceInfo();
-					bool open(uint32_t _device,
-					          enum audio::orchestra::mode _mode,
-					          uint32_t _channels,
-					          uint32_t _firstChannel,
-					          uint32_t _sampleRate,
-					          enum audio::format _format,
-					          uint32_t *_bufferSize,
-					          const audio::orchestra::StreamOptions& _options);
-					
-					bool openName(const etk::String& _deviceName,
-					              audio::orchestra::mode _mode,
-					              uint32_t _channels,
-					              uint32_t _firstChannel,
-					              uint32_t _sampleRate,
-					              audio::format _format,
-					              uint32_t *_bufferSize,
-					              const audio::orchestra::StreamOptions& _options);
-					virtual audio::Time getStreamTime();
-				public:
-					bool isMasterOf(ememory::SharedPtr<audio::orchestra::Api> _api);
-			};
-		}
-	}
-}
-
-#endif

audio/orchestra/api/Android.cpp

@@ -1,195 +0,0 @@
-/** @file
- * @author Edouard DUPIN 
- * @copyright 2011, Edouard DUPIN, all right reserved
- * @license APACHE v2.0 (see license file)
- * @fork from RTAudio
- */
-
-#ifdef ORCHESTRA_BUILD_JAVA
-
-//#include <ewol/context/Context.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>(ETK_NEW(audio::orchestra::api::Android));
-}
-
-
-audio::orchestra::api::Android::Android() :
-	m_uid(-1) {
-	ATA_INFO("Create Android interface");
-}
-
-audio::orchestra::api::Android::~Android() {
-	ATA_INFO("Destroy Android interface");
-}
-
-uint32_t audio::orchestra::api::Android::getDeviceCount() {
-	//ATA_INFO("Get device count:"<< m_devices.size());
-	return audio::orchestra::api::android::getDeviceCount();
-}
-
-audio::orchestra::DeviceInfo audio::orchestra::api::Android::getDeviceInfo(uint32_t _device) {
-	//ATA_INFO("Get device info ...");
-	return audio::orchestra::api::android::getDeviceInfo(_device);
-}
-
-enum audio::orchestra::error audio::orchestra::api::Android::closeStream() {
-	ATA_INFO("Close Stream");
-	// Can not close the stream now...
-	return audio::orchestra::error_none;
-}
-
-enum audio::orchestra::error audio::orchestra::api::Android::startStream() {
-	ATA_INFO("Start Stream");
-	// TODO : Check return ...
-	audio::orchestra::Api::startStream();
-	// Can not close the stream now...
-	return audio::orchestra::api::android::startStream(m_uid);
-}
-
-enum audio::orchestra::error audio::orchestra::api::Android::stopStream() {
-	ATA_INFO("Stop stream");
-	// Can not close the stream now...
-	return audio::orchestra::api::android::stopStream(m_uid);
-}
-
-enum audio::orchestra::error audio::orchestra::api::Android::abortStream() {
-	ATA_INFO("Abort Stream");
-	// Can not close the stream now...
-	return audio::orchestra::error_none;
-}
-
-
-void audio::orchestra::api::Android::playback(int16_t* _dst, int32_t _nbChunk) {
-	// clear output buffer:
-	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();
-	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(null,
-		                          audio::Time(),
-		                          &m_userBuffer[m_mode][0],
-		                          streamTime,
-		                          uint32_t(_nbChunk),
-		                          status);
-		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(null,
-		                          audio::Time(),
-		                          _dst,
-		                          streamTime,
-		                          uint32_t(_nbChunk),
-		                          status);
-		
-	}
-	if (doStopStream == 2) {
-		abortStream();
-		return;
-	}
-	audio::orchestra::Api::tickStreamTime();
-}
-
-void audio::orchestra::api::Android::record(int16_t* _dst, int32_t _nbChunk) {
-	int32_t doStopStream = 0;
-	audio::Time streamTime = getStreamTime();
-	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,
-		                          null,
-		                          audio::Time(),
-		                          uint32_t(_nbChunk),
-		                          status);
-	} else {
-		ATA_VERBOSE("Need playback data " << int32_t(_nbChunk) << " pointer=" << int64_t(_dst));
-		doStopStream = m_callback(_dst,
-		                          streamTime,
-		                          null,
-		                          audio::Time(),
-		                          uint32_t(_nbChunk),
-		                          status);
-		
-	}
-	if (doStopStream == 2) {
-		abortStream();
-		return;
-	}
-	audio::orchestra::Api::tickStreamTime();
-}
-
-bool audio::orchestra::api::Android::open(uint32_t _device,
-                                          audio::orchestra::mode _mode,
-                                          uint32_t _channels,
-                                          uint32_t _firstChannel,
-                                          uint32_t _sampleRate,
-                                          audio::format _format,
-                                          uint32_t *_bufferSize,
-                                          const audio::orchestra::StreamOptions& _options) {
-	bool ret = false;
-	ATA_INFO("Probe : device=" << _device << " channels=" << _channels << " firstChannel=" << _firstChannel << " sampleRate=" << _sampleRate);
-	m_mode = _mode;
-	m_userFormat = _format;
-	m_nUserChannels[modeToIdTable(m_mode)] = _channels;
-	
-	m_uid = audio::orchestra::api::android::open(_device, m_mode, _channels, _firstChannel, _sampleRate, _format, _bufferSize, _options, ememory::staticPointerCast<audio::orchestra::api::Android>(sharedFromThis()));
-	if (m_uid < 0) {
-		ret = false;
-	} else {
-		ret = true;
-	}
-	m_bufferSize = 256;
-	m_sampleRate = _sampleRate;
-	m_doByteSwap[modeToIdTable(m_mode)] = false; // for endienness ...
-	
-	// TODO : For now, we write it in hard ==> to bu update later ...
-	m_deviceFormat[modeToIdTable(m_mode)] = audio::format_int16;
-	m_nDeviceChannels[modeToIdTable(m_mode)] = 2;
-	m_deviceInterleaved[modeToIdTable(m_mode)] = true;
-	
-	m_doConvertBuffer[modeToIdTable(m_mode)] = false;
-	if (m_userFormat != m_deviceFormat[modeToIdTable(m_mode)]) {
-		m_doConvertBuffer[modeToIdTable(m_mode)] = true;
-	}
-	if (m_nUserChannels[modeToIdTable(m_mode)] < m_nDeviceChannels[modeToIdTable(m_mode)]) {
-		m_doConvertBuffer[modeToIdTable(m_mode)] = true;
-	}
-	if (    m_deviceInterleaved[modeToIdTable(m_mode)] == false
-	     && m_nUserChannels[modeToIdTable(m_mode)] > 1) {
-		m_doConvertBuffer[modeToIdTable(m_mode)] = true;
-	}
-	if (m_doConvertBuffer[modeToIdTable(m_mode)] == true) {
-		// Allocate necessary internal buffers.
-		uint64_t bufferBytes = m_nUserChannels[modeToIdTable(m_mode)] * m_bufferSize * audio::getFormatBytes(m_userFormat);
-		m_userBuffer[modeToIdTable(m_mode)].resize(bufferBytes);
-		if (m_userBuffer[modeToIdTable(m_mode)].size() == 0) {
-			ATA_ERROR("error allocating user buffer memory.");
-		}
-		setConvertInfo(m_mode, _firstChannel);
-	}
-	ATA_INFO("device format : " << m_deviceFormat[modeToIdTable(m_mode)] << " user format : " << m_userFormat);
-	ATA_INFO("device channels : " << m_nDeviceChannels[modeToIdTable(m_mode)] << " user channels : " << m_nUserChannels[modeToIdTable(m_mode)]);
-	ATA_INFO("do convert buffer : " << m_doConvertBuffer[modeToIdTable(m_mode)]);
-	if (ret == false) {
-		ATA_ERROR("Can not open device.");
-	}
-	return ret;
-}
-
-#endif
-

audio/orchestra/api/Android.hpp

@@ -1,60 +0,0 @@
-/** @file
- * @author Edouard DUPIN 
- * @copyright 2011, Edouard DUPIN, all right reserved
- * @license APACHE v2.0 (see license file)
- */
-#pragma once
-
-#ifdef ORCHESTRA_BUILD_JAVA
-
-#include <audio/orchestra/Interface.hpp>
-
-namespace audio {
-	namespace orchestra {
-		namespace api {
-			class Android: public audio::orchestra::Api {
-				public:
-					static ememory::SharedPtr<audio::orchestra::Api> create();
-				public:
-					Android();
-					virtual ~Android();
-					const etk::String& getCurrentApi() {
-						return audio::orchestra::typeJava;
-					}
-					uint32_t getDeviceCount();
-					audio::orchestra::DeviceInfo getDeviceInfo(uint32_t _device);
-					enum audio::orchestra::error closeStream();
-					enum audio::orchestra::error startStream();
-					enum audio::orchestra::error stopStream();
-					enum audio::orchestra::error abortStream();
-					// This function is intended for internal use only.	It must be
-					// public because it is called by the internal callback handler,
-					// which is not a member of RtAudio.	External use of this function
-					// will most likely produce highly undesireable results!
-					void callbackEvent();
-				private:
-					int32_t m_uid;
-				public:
-					int32_t getUId() {
-						return m_uid;
-					}
-				private:
-					etk::Vector<audio::orchestra::DeviceInfo> m_devices;
-					void saveDeviceInfo();
-					bool open(uint32_t _device,
-					          audio::orchestra::mode _mode,
-					          uint32_t _channels,
-					          uint32_t _firstChannel,
-					          uint32_t _sampleRate,
-					          audio::format _format,
-					          uint32_t *_bufferSize,
-					          const audio::orchestra::StreamOptions& _options);
-				public:
-					void playback(int16_t* _dst, int32_t _nbChunk);
-					void record(int16_t* _dst, int32_t _nbChunk);
-			};
-		}
-	}
-}
-
-#endif

audio/orchestra/api/AndroidNativeInterface.cpp

@@ -1,542 +0,0 @@
-/** @file
- * @author Edouard DUPIN 
- * @copyright 2011, Edouard DUPIN, all right reserved
- * @license APACHE v2.0 (see license file)
- */
-
-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.hpp>
-#include <ememory/memory.hpp>
-#include <ejson/ejson.hpp>
-
-class AndroidOrchestraContext {
-	public:
-		// get a resources from the java environement : 
-		JNIEnv* m_JavaVirtualMachinePointer; //!< the JVM
-		jclass m_javaClassOrchestra; //!< main activity class (android ...)
-		jclass m_javaClassOrchestraCallback;
-		jobject m_javaObjectOrchestraCallback;
-		jmethodID m_javaMethodOrchestraActivityAudioGetDeviceCount;
-		jmethodID m_javaMethodOrchestraActivityAudioGetDeviceProperty;
-		jmethodID m_javaMethodOrchestraActivityAudioOpenDeviceInput;
-		jmethodID m_javaMethodOrchestraActivityAudioOpenDeviceOutput;
-		jmethodID m_javaMethodOrchestraActivityAudioCloseDevice;
-		jmethodID m_javaMethodOrchestraActivityAudioStart;
-		jmethodID m_javaMethodOrchestraActivityAudioStop;
-		jclass m_javaDefaultClassString; //!< default string class
-	private:
-		bool safeInitMethodID(jmethodID& _mid, jclass& _cls, const char* _name, const char* _sign) {
-			_mid = m_JavaVirtualMachinePointer->GetMethodID(_cls, _name, _sign);
-			if(_mid == null) {
-				ATA_ERROR("C->java : Can't find the method " << _name);
-				/* remove access on the virtual machine : */
-				m_JavaVirtualMachinePointer = null;
-				return false;
-			}
-			return true;
-		}
-		bool java_attach_current_thread(int *_rstatus) {
-			ATA_DEBUG("C->java : call java");
-			if (jvm_basics::getJavaVM() == null) {
-				ATA_ERROR("C->java : JVM not initialised");
-				m_JavaVirtualMachinePointer = null;
-				return false;
-			}
-			*_rstatus = jvm_basics::getJavaVM()->GetEnv((void **) &m_JavaVirtualMachinePointer, JNI_VERSION_1_6);
-			if (*_rstatus == JNI_EDETACHED) {
-				JavaVMAttachArgs lJavaVMAttachArgs;
-				lJavaVMAttachArgs.version = JNI_VERSION_1_6;
-				lJavaVMAttachArgs.name = "EwolNativeThread";
-				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 = null;
-					return false;
-				}
-			}
-			return true;
-		}
-		void java_detach_current_thread(int _status) {
-			if(_status == JNI_EDETACHED) {
-				jvm_basics::getJavaVM()->DetachCurrentThread();
-				m_JavaVirtualMachinePointer = null;
-			}
-		}
-		
-	public:
-		AndroidOrchestraContext(JNIEnv* _env, jclass _classBase, jobject _objCallback) :
-		  m_JavaVirtualMachinePointer(null),
-		  m_javaClassOrchestra(0),
-		  m_javaClassOrchestraCallback(0),
-		  m_javaObjectOrchestraCallback(0),
-		  
-		  m_javaMethodOrchestraActivityAudioGetDeviceCount(0),
-		  m_javaMethodOrchestraActivityAudioGetDeviceProperty(0),
-		  m_javaMethodOrchestraActivityAudioOpenDeviceInput(0),
-		  m_javaMethodOrchestraActivityAudioOpenDeviceOutput(0),
-		  m_javaMethodOrchestraActivityAudioCloseDevice(0),
-		  m_javaMethodOrchestraActivityAudioStart(0),
-		  m_javaMethodOrchestraActivityAudioStop(0),
-		  m_javaDefaultClassString(0) {
-			ATA_DEBUG("*******************************************");
-			ATA_DEBUG("** set JVM Pointer (orchestra)           **");
-			ATA_DEBUG("*******************************************");
-			m_JavaVirtualMachinePointer = _env;
-			// get default needed all time elements : 
-			if (m_JavaVirtualMachinePointer == null) {
-				ATA_ERROR("C->java: NULLPTR jvm interface");
-				return;
-			}
-			ATA_DEBUG("C->java: try load org/musicdsp/orchestra/OrchestraNative class");
-			m_javaClassOrchestra = m_JavaVirtualMachinePointer->FindClass("org/musicdsp/orchestra/OrchestraNative" );
-			if (m_javaClassOrchestra == 0) {
-				ATA_ERROR("C->java : Can't find org/musicdsp/orchestra/OrchestraNative class");
-				// remove access on the virtual machine : 
-				m_JavaVirtualMachinePointer = null;
-				return;
-			}
-			/* The object field extends Activity and implement OrchestraCallback */
-			m_javaClassOrchestraCallback = m_JavaVirtualMachinePointer->GetObjectClass(_objCallback);
-			if(m_javaClassOrchestraCallback == null) {
-				ATA_ERROR("C->java : Can't find org/musicdsp/orchestra/OrchestraManagerCallback class");
-				// remove access on the virtual machine : 
-				m_JavaVirtualMachinePointer = null;
-				return;
-			}
-			bool functionCallbackIsMissing = false;
-			bool ret= false;
-			
-			ret = safeInitMethodID(m_javaMethodOrchestraActivityAudioGetDeviceCount,
-			                       m_javaClassOrchestraCallback,
-			                       "getDeviceCount",
-			                       "()I");
-			if (ret == false) {
-				jvm_basics::checkExceptionJavaVM(_env);
-				ATA_ERROR("system can not start without function : getDeviceCount");
-				functionCallbackIsMissing = true;
-			}
-			
-			ret = safeInitMethodID(m_javaMethodOrchestraActivityAudioGetDeviceProperty,
-			                       m_javaClassOrchestraCallback,
-			                       "getDeviceProperty",
-			                       "(I)Ljava/lang/String;");
-			if (ret == false) {
-				jvm_basics::checkExceptionJavaVM(_env);
-				ATA_ERROR("system can not start without function : getDeviceProperty");
-				functionCallbackIsMissing = true;
-			}
-			ret = safeInitMethodID(m_javaMethodOrchestraActivityAudioOpenDeviceInput,
-			                       m_javaClassOrchestraCallback,
-			                       "openDeviceInput",
-			                       "(IIII)I");
-			if (ret == false) {
-				jvm_basics::checkExceptionJavaVM(_env);
-				ATA_ERROR("system can not start without function : openDeviceInput");
-				functionCallbackIsMissing = true;
-			}
-			ret = safeInitMethodID(m_javaMethodOrchestraActivityAudioOpenDeviceOutput,
-			                       m_javaClassOrchestraCallback,
-			                       "openDeviceOutput",
-			                       "(IIII)I");
-			if (ret == false) {
-				jvm_basics::checkExceptionJavaVM(_env);
-				ATA_ERROR("system can not start without function : openDeviceOutput");
-				functionCallbackIsMissing = true;
-			}
-			ret = safeInitMethodID(m_javaMethodOrchestraActivityAudioCloseDevice,
-			                       m_javaClassOrchestraCallback,
-			                       "closeDevice",
-			                       "(I)Z");
-			if (ret == false) {
-				jvm_basics::checkExceptionJavaVM(_env);
-				ATA_ERROR("system can not start without function : closeDevice");
-				functionCallbackIsMissing = true;
-			}
-			ret = safeInitMethodID(m_javaMethodOrchestraActivityAudioStart,
-			                       m_javaClassOrchestraCallback,
-			                       "start",
-			                       "(I)Z");
-			if (ret == false) {
-				jvm_basics::checkExceptionJavaVM(_env);
-				ATA_ERROR("system can not start without function : start");
-				functionCallbackIsMissing = true;
-			}
-			ret = safeInitMethodID(m_javaMethodOrchestraActivityAudioStop,
-			                       m_javaClassOrchestraCallback,
-			                       "stop",
-			                       "(I)Z");
-			if (ret == false) {
-				jvm_basics::checkExceptionJavaVM(_env);
-				ATA_ERROR("system can not start without function : stop");
-				functionCallbackIsMissing = true;
-			}
-			
-			
-			m_javaObjectOrchestraCallback = _env->NewGlobalRef(_objCallback);
-			if (m_javaObjectOrchestraCallback == null) {
-				functionCallbackIsMissing = true;
-			}
-			
-			m_javaDefaultClassString = m_JavaVirtualMachinePointer->FindClass("java/lang/String" );
-			if (m_javaDefaultClassString == 0) {
-				ATA_ERROR("C->java : Can't find java/lang/String" );
-				// remove access on the virtual machine : 
-				m_JavaVirtualMachinePointer = null;
-				functionCallbackIsMissing = true;
-			}
-			if (functionCallbackIsMissing == true) {
-				ATA_CRITICAL(" mission one function ==> system can not work withut it...");
-			}
-		}
-		
-		~AndroidOrchestraContext() {
-			// TODO ...
-		}
-		
-		void unInit(JNIEnv* _env) {
-			_env->DeleteGlobalRef(m_javaObjectOrchestraCallback);
-			m_javaObjectOrchestraCallback = null;
-		}
-		
-		uint32_t getDeviceCount() {
-			// Request the clipBoard :
-			ATA_WARNING("C->java : audio get device count");
-			int status;
-			if(!java_attach_current_thread(&status)) {
-				return 0;
-			}
-			ATA_DEBUG("Call CallIntMethod ...");
-			//Call java ...
-			jint ret = m_JavaVirtualMachinePointer->CallIntMethod(m_javaObjectOrchestraCallback, m_javaMethodOrchestraActivityAudioGetDeviceCount);
-			// manage execption : 
-			jvm_basics::checkExceptionJavaVM(m_JavaVirtualMachinePointer);
-			java_detach_current_thread(status);
-			ATA_WARNING(" find " << (uint32_t)ret << " IO");
-			return (uint32_t)ret;
-		}
-		audio::orchestra::DeviceInfo getDeviceInfo(uint32_t _idDevice) {
-			audio::orchestra::DeviceInfo info;
-			// Request the clipBoard :
-			ATA_WARNING("C->java : audio get device info " << _idDevice);
-			int status;
-			if(!java_attach_current_thread(&status)) {
-				return info;
-			}
-			//Call java ...
-			jstring returnString = (jstring) m_JavaVirtualMachinePointer->CallObjectMethod(m_javaObjectOrchestraCallback, m_javaMethodOrchestraActivityAudioGetDeviceProperty, _idDevice);
-			const char *js = m_JavaVirtualMachinePointer->GetStringUTFChars(returnString, null);
-			etk::String retString(js);
-			m_JavaVirtualMachinePointer->ReleaseStringUTFChars(returnString, js);
-			//m_JavaVirtualMachinePointer->DeleteLocalRef(returnString);
-			// manage execption : 
-			jvm_basics::checkExceptionJavaVM(m_JavaVirtualMachinePointer);
-			java_detach_current_thread(status);
-			ATA_WARNING("get device information : " << retString);
-			ejson::Document doc;
-			if (doc.parse(retString) == false) {
-				
-				return info;
-			}
-			info.name = doc["name"].toString().get("no-name");
-			if (doc["type"].toString().get("output") == "output") {
-				info.input = false;
-			} else {
-				info.input = true;
-			}
-			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["channels"].toArray();
-			if (list.exist() == true) {
-				for (auto it : list) {
-					info.channels.pushBack(audio::getChannelFromString(it.toString().get("???")));
-				}
-			}
-			list = doc["format"].toArray();
-			if (list.exist() == true) {
-				for (auto it : list) {
-					info.nativeFormats.pushBack(audio::getFormatFromString(it.toString().get("???")));
-				}
-			}
-			info.isDefault = doc["default"].toBoolean().get(false);
-			info.isCorrect = true;
-			return info;
-		}
-	private:
-		etk::Vector<ememory::WeakPtr<audio::orchestra::api::Android> > m_instanceList; // list of connected handle ...
-		//AndroidAudioCallback m_audioCallBack;
-		//void* m_audioCallBackUserData;
-	public:
-		int32_t open(uint32_t _idDevice,
-		             audio::orchestra::mode _mode,
-		             uint32_t _channels,
-		             uint32_t _firstChannel,
-		             uint32_t _sampleRate,
-		             audio::format _format,
-		             uint32_t *_bufferSize,
-		             const audio::orchestra::StreamOptions& _options,
-		             ememory::SharedPtr<audio::orchestra::api::Android> _instance) {
-			ATA_DEBUG("C->java : audio open device");
-			int status;
-			if(!java_attach_current_thread(&status)) {
-				return -1;
-			}
-			//Call java ...
-			jint ret = false;
-			if (_mode == audio::orchestra::mode_output) {
-				ret = m_JavaVirtualMachinePointer->CallIntMethod(m_javaObjectOrchestraCallback, m_javaMethodOrchestraActivityAudioOpenDeviceOutput, _idDevice, _sampleRate, _channels, /*_format*/ 1);
-			} else {
-				ret = m_JavaVirtualMachinePointer->CallIntMethod(m_javaObjectOrchestraCallback, m_javaMethodOrchestraActivityAudioOpenDeviceInput, _idDevice, _sampleRate, _channels, /*_format*/ 1);
-			}
-			// manage execption : 
-			jvm_basics::checkExceptionJavaVM(m_JavaVirtualMachinePointer);
-			java_detach_current_thread(status);
-			if (int32_t(ret) >= 0) {
-				m_instanceList.pushBack(_instance);
-				return int32_t(ret);
-			}
-			return -1;
-		}
-	public:
-		enum audio::orchestra::error closeStream(int32_t _id) {
-			ATA_DEBUG("C->java : audio close device");
-			int status;
-			if(!java_attach_current_thread(&status)) {
-				return audio::orchestra::error_fail;
-			}
-			//Call java ...
-			jboolean ret = m_JavaVirtualMachinePointer->CallBooleanMethod(m_javaObjectOrchestraCallback, m_javaMethodOrchestraActivityAudioCloseDevice, _id);
-			// manage execption : 
-			jvm_basics::checkExceptionJavaVM(m_JavaVirtualMachinePointer);
-			java_detach_current_thread(status);
-			if (bool(ret) == false) {
-				return audio::orchestra::error_fail;
-			}
-			return audio::orchestra::error_none;
-		}
-		
-		enum audio::orchestra::error startStream(int32_t _id) {
-			ATA_DEBUG("C->java : audio start device");
-			int status;
-			if(!java_attach_current_thread(&status)) {
-				return audio::orchestra::error_fail;
-			}
-			//Call java ...
-			jboolean ret = m_JavaVirtualMachinePointer->CallBooleanMethod(m_javaObjectOrchestraCallback, m_javaMethodOrchestraActivityAudioStart, _id);
-			// manage execption : 
-			jvm_basics::checkExceptionJavaVM(m_JavaVirtualMachinePointer);
-			java_detach_current_thread(status);
-			if (bool(ret) == false) {
-				return audio::orchestra::error_fail;
-			}
-			return audio::orchestra::error_none;
-		}
-		enum audio::orchestra::error stopStream(int32_t _id) {
-			
-			ATA_DEBUG("C->java : audio close device");
-			int status;
-			if(!java_attach_current_thread(&status)) {
-				return audio::orchestra::error_fail;
-			}
-			//Call java ...
-			jboolean ret = m_JavaVirtualMachinePointer->CallBooleanMethod(m_javaObjectOrchestraCallback, m_javaMethodOrchestraActivityAudioStop, _id);
-			// manage execption : 
-			jvm_basics::checkExceptionJavaVM(m_JavaVirtualMachinePointer);
-			java_detach_current_thread(status);
-			if (bool(ret) == false) {
-				return audio::orchestra::error_fail;
-			}
-			return audio::orchestra::error_none;
-		}
-		enum audio::orchestra::error abortStream(int32_t _id) {
-			return audio::orchestra::error_fail;
-		}
-		
-		void playback(int32_t _id, int16_t* _dst, int32_t _nbChunk) {
-			auto it = m_instanceList.begin();
-			while (it != m_instanceList.end()) {
-				auto elem = it->lock();
-				if (elem == null) {
-					it = m_instanceList.erase(it);
-					continue;
-				}
-				if (elem->getUId() == _id) {
-					elem->playback(_dst, _nbChunk);
-				}
-				++it;
-			}
-		}
-		void record(int32_t _id, int16_t* _dst, int32_t _nbChunk) {
-			auto it = m_instanceList.begin();
-			while (it != m_instanceList.end()) {
-				auto elem = it->lock();
-				if (elem == null) {
-					it = m_instanceList.erase(it);
-					continue;
-				}
-				if (elem->getUId() == _id) {
-					elem->record(_dst, _nbChunk);
-				}
-				++it;
-			}
-		}
-};
-
-static ememory::SharedPtr<AndroidOrchestraContext> s_localContext;
-static int32_t s_nbContextRequested(0);
-
-
-uint32_t audio::orchestra::api::android::getDeviceCount() {
-	if (s_localContext == null) {
-		ATA_ERROR("Have no Orchertra API instanciate in JAVA ...");
-		return 0;
-	}
-	return s_localContext->getDeviceCount();
-}
-
-audio::orchestra::DeviceInfo audio::orchestra::api::android::getDeviceInfo(uint32_t _device) {
-	if (s_localContext == null) {
-		return audio::orchestra::DeviceInfo();
-	}
-	return s_localContext->getDeviceInfo(_device);
-}
-
-int32_t audio::orchestra::api::android::open(uint32_t _device,
-                                             audio::orchestra::mode _mode,
-                                             uint32_t _channels,
-                                             uint32_t _firstChannel,
-                                             uint32_t _sampleRate,
-                                             audio::format _format,
-                                             uint32_t *_bufferSize,
-                                             const audio::orchestra::StreamOptions& _options,
-                                             ememory::SharedPtr<audio::orchestra::api::Android> _instance) {
-	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 == 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 == 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 == 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 == null) {
-		return audio::orchestra::error_fail;
-	}
-	return s_localContext->abortStream(_id);
-}
-
-extern "C" {
-	void Java_org_musicdsp_orchestra_OrchestraNative_NNsetJavaManager(JNIEnv* _env,
-	                                                                  jclass _classBase,
-	                                                                  jobject _objCallback) {
-		ethread::UniqueLock lock(jvm_basics::getMutexJavaVM());
-		ATA_INFO("*******************************************");
-		ATA_INFO("** Creating Orchestra context            **");
-		ATA_INFO("*******************************************");
-		if (s_localContext != null) {
-			s_nbContextRequested++;
-		}
-		s_localContext = ememory::makeShared<AndroidOrchestraContext>(_env, _classBase, _objCallback);
-		if (s_localContext == null) {
-			ATA_ERROR("Can not allocate the orchestra main context instance");
-			return;
-		}
-		s_nbContextRequested++;
-	}
-	
-	void Java_org_musicdsp_orchestra_OrchestraNative_NNsetJavaManagerRemove(JNIEnv* _env, jclass _cls) {
-		ethread::UniqueLock lock(jvm_basics::getMutexJavaVM());
-		ATA_INFO("*******************************************");
-		ATA_INFO("** remove Orchestra Pointer              **");
-		ATA_INFO("*******************************************");
-		if (s_nbContextRequested == 0) {
-			ATA_ERROR("Request remove orchestra interface from Android, but no more interface availlable");
-			return;
-		}
-		s_nbContextRequested--;
-		if (s_nbContextRequested == 0) {
-			s_localContext.reset();
-		}
-	}
-	void Java_org_musicdsp_orchestra_OrchestraNative_NNPlayback(JNIEnv* _env,
-	                                                            void* _reserved,
-	                                                            jint _id,
-	                                                            jshortArray _location,
-	                                                            jint _nbChunk) {
-		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 != null) {
-			//ATA_INFO("Need audioData " << int32_t(_nbChunk));
-			s_localContext->playback(int32_t(_id), static_cast<short*>(dst), int32_t(_nbChunk));
-		}
-		// TODO : Understand why it did not work corectly ...
-		//if (isCopy == JNI_TRUE) {
-		// release the short* pointer
-		_env->ReleaseShortArrayElements(_location, dst, 0);
-		//}
-	}
-	void Java_org_musicdsp_orchestra_OrchestraNative_NNRecord(JNIEnv* _env,
-	                                                          void* _reserved,
-	                                                          jint _id,
-	                                                          jshortArray _location,
-	                                                          jint _nbChunk) {
-		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 != null) {
-			//ATA_INFO("Need audioData " << int32_t(_nbChunk));
-			s_localContext->record(int32_t(_id), static_cast<short*>(dst), int32_t(_nbChunk));
-		}
-		// TODO : Understand why it did not work corectly ...
-		//if (isCopy == JNI_TRUE) {
-		// release the short* pointer
-		_env->ReleaseShortArrayElements(_location, dst, 0);
-		//}
-	}
-}
-

audio/orchestra/api/AndroidNativeInterface.hpp

@@ -1,42 +0,0 @@
-/** @file
- * @author Edouard DUPIN 
- * @copyright 2011, Edouard DUPIN, all right reserved
- * @license APACHE v2.0 (see license file)
- */
-#pragma once
-
-#ifdef ORCHESTRA_BUILD_JAVA
-
-#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 {
-		namespace api {
-			class Android;
-			namespace android {
-				uint32_t getDeviceCount();
-				audio::orchestra::DeviceInfo getDeviceInfo(uint32_t _device);
-				int32_t open(uint32_t _device,
-				             audio::orchestra::mode _mode,
-				             uint32_t _channels,
-				             uint32_t _firstChannel,
-				             uint32_t _sampleRate,
-				             audio::format _format,
-				             uint32_t *_bufferSize,
-				             const audio::orchestra::StreamOptions& _options,
-				             ememory::SharedPtr<audio::orchestra::api::Android> _instance);
-				enum audio::orchestra::error closeStream(int32_t _id);
-				enum audio::orchestra::error startStream(int32_t _id);
-				enum audio::orchestra::error stopStream(int32_t _id);
-				enum audio::orchestra::error abortStream(int32_t _id);
-			}
-		}
-	}
-}
-
-#endif

audio/orchestra/api/Asio.cpp

@@ -1,921 +0,0 @@
-/** @file
- * @author Edouard DUPIN 
- * @copyright 2011, Edouard DUPIN, all right reserved
- * @license APACHE v2.0 (see license file)
- * @fork from RTAudio
- */
-
-
-#if defined(ORCHESTRA_BUILD_ASIO)
-
-#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>(ETK_NEW(audio::orchestra::api::Asio));
-}
-
-
-// The ASIO API is designed around a callback scheme, so this
-// implementation is similar to that used for OS-X CoreAudio and Linux
-// Jack.	The primary constraint with ASIO is that it only allows
-// access to a single driver at a time.	Thus, it is not possible to
-// have more than one simultaneous RtAudio stream.
-//
-// This implementation also requires a number of external ASIO files
-// and a few global variables.	The ASIO callback scheme does not
-// allow for the passing of user data, so we must create a global
-// pointer to our callbackInfo structure.
-//
-// On unix systems, we make use of a pthread condition variable.
-// Since there is no equivalent in Windows, I hacked something based
-// on information found in
-// http://www.cs.wustl.edu/~schmidt/win32-cv-1.html.
-
-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;
-static CallbackInfo *asioCallbackInfo;
-static bool asioXRun;
-
-namespace audio {
-	namespace orchestra {
-		namespace api {
-			class AsioPrivate {
-				public:
-					int32_t drainCounter; // Tracks callback counts when draining
-					bool internalDrain; // Indicates if stop is initiated from callback or not.
-					ASIOBufferInfo *bufferInfos;
-					HANDLE condition;
-					AsioPrivate() :
-					  drainCounter(0),
-					  internalDrain(false),
-					  bufferInfos(0) {
-						
-					}
-			};
-		}
-	}
-}
-
-// Function declarations (definitions at end of section)
-static const char* getAsioErrorString(ASIOError _result);
-static void sampleRateChanged(ASIOSampleRate _sRate);
-static long asioMessages(long _selector, long _value, void* _message, double* _opt);
-
-audio::orchestra::api::Asio::Asio() :
-  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(null); 
-	if (FAILED(hr)) {
-		ATA_ERROR("requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)");
-	}
-	m_coInitialized = true;
-	drivers.removeCurrentDriver();
-	driverInfo.asioVersion = 2;
-	// See note in DirectSound implementation about GetDesktopWindow().
-	driverInfo.sysRef = GetForegroundWindow();
-}
-
-audio::orchestra::api::Asio::~Asio() {
-	if (m_state != audio::orchestra::state::closed) {
-		closeStream();
-	}
-	if (m_coInitialized) {
-		CoUninitialize();
-	}
-}
-
-uint32_t audio::orchestra::api::Asio::getDeviceCount() {
-	return (uint32_t) drivers.asioGetNumDev();
-}
-
-rtaudio::DeviceInfo audio::orchestra::api::Asio::getDeviceInfo(uint32_t _device) {
-	rtaudio::DeviceInfo info;
-	info.probed = false;
-	// Get device ID
-	uint32_t nDevices = getDeviceCount();
-	if (nDevices == 0) {
-		ATA_ERROR("no devices found!");
-		return info;
-	}
-	if (_device >= nDevices) {
-		ATA_ERROR("device ID is invalid!");
-		return info;
-	}
-	// If a stream is already open, we cannot probe other devices.	Thus, use the saved results.
-	if (m_state != audio::orchestra::state::closed) {
-		if (_device >= m_devices.size()) {
-			ATA_ERROR("device ID was not present before stream was opened.");
-			return info;
-		}
-		return m_devices[ _device ];
-	}
-	char driverName[32];
-	ASIOError result = drivers.asioGetDriverName((int) _device, driverName, 32);
-	if (result != ASE_OK) {
-		ATA_ERROR("unable to get driver name (" << getAsioErrorString(result) << ").");
-		return info;
-	}
-	info.name = driverName;
-	if (!drivers.loadDriver(driverName)) {
-		ATA_ERROR("unable to load driver (" << driverName << ").");
-		return info;
-	}
-	result = ASIOInit(&driverInfo);
-	if (result != ASE_OK) {
-		ATA_ERROR("error (" << getAsioErrorString(result) << ") initializing driver (" << driverName << ").");
-		return info;
-	}
-	// Determine the device channel information.
-	long inputChannels, outputChannels;
-	result = ASIOGetChannels(&inputChannels, &outputChannels);
-	if (result != ASE_OK) {
-		drivers.removeCurrentDriver();
-		ATA_ERROR("error (" << getAsioErrorString(result) << ") getting channel count (" << driverName << ").");
-		return info;
-	}
-	info.outputChannels = outputChannels;
-	info.inputChannels = inputChannels;
-	if (info.outputChannels > 0 && info.inputChannels > 0) {
-		info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
-	}
-	// Determine the supported sample rates.
-	info.sampleRates.clear();
-	for (uint32_t i=0; i<MAX_SAMPLE_RATES; i++) {
-		result = ASIOCanSampleRate((ASIOSampleRate) SAMPLE_RATES[i]);
-		if (result == ASE_OK) {
-			info.sampleRates.pushBack(SAMPLE_RATES[i]);
-		}
-	}
-	// Determine supported data types ... just check first channel and assume rest are the same.
-	ASIOChannelInfo channelInfo;
-	channelInfo.channel = 0;
-	channelInfo.isInput = true;
-	if (info.inputChannels <= 0) {
-		channelInfo.isInput = false;
-	}
-	result = ASIOGetChannelInfo(&channelInfo);
-	if (result != ASE_OK) {
-		drivers.removeCurrentDriver();
-		ATA_ERROR("error (" << getAsioErrorString(result) << ") getting driver channel info (" << driverName << ").");
-		return info;
-	}
-	info.nativeFormats.clear();
-	if (    channelInfo.type == ASIOSTInt16MSB
-	     || channelInfo.type == ASIOSTInt16LSB) {
-		info.nativeFormats.pushBack(audio::format_int16);
-	} else if (    channelInfo.type == ASIOSTInt32MSB
-	            || channelInfo.type == ASIOSTInt32LSB) {
-		info.nativeFormats.pushBack(audio::format_int32);
-	} else if (    channelInfo.type == ASIOSTFloat32MSB
-	            || channelInfo.type == ASIOSTFloat32LSB) {
-		info.nativeFormats.pushBack(audio::format_float);
-	} else if (    channelInfo.type == ASIOSTFloat64MSB
-	            || channelInfo.type == ASIOSTFloat64LSB) {
-		info.nativeFormats.pushBack(audio::format_double);
-	} else if (    channelInfo.type == ASIOSTInt24MSB
-	            || channelInfo.type == ASIOSTInt24LSB) {
-		info.nativeFormats.pushBack(audio::format_int24);
-	}
-	if (info.outputChannels > 0){
-		if (getDefaultOutputDevice() == _device) {
-			info.isDefaultOutput = true;
-		}
-	}
-	if (info.inputChannels > 0) {
-		if (getDefaultInputDevice() == _device) {
-			info.isDefaultInput = true;
-		}
-	}
-	info.probed = true;
-	drivers.removeCurrentDriver();
-	return info;
-}
-
-static void bufferSwitch(long _index, ASIOBool _processNow) {
-	RtApiAsio* object = (RtApiAsio*)asioCallbackInfo->object;
-	object->callbackEvent(_index);
-}
-
-void audio::orchestra::api::Asio::saveDeviceInfo() {
-	m_devices.clear();
-	uint32_t nDevices = getDeviceCount();
-	m_devices.resize(nDevices);
-	for (uint32_t i=0; i<nDevices; i++) {
-		m_devices[i] = getDeviceInfo(i);
-	}
-}
-
-bool audio::orchestra::api::Asio::open(uint32_t _device,
-                                       audio::orchestra::mode _mode,
-                                       uint32_t _channels,
-                                       uint32_t _firstChannel,
-                                       uint32_t _sampleRate,
-                                       audio::format _format,
-                                       uint32_t* _bufferSize,
-                                       const audio::orchestra::StreamOptions& _options) {
-	// For ASIO, a duplex stream MUST use the same driver.
-	if (    _mode == audio::orchestra::mode_input
-	     && m_mode == audio::orchestra::mode_output
-	     && m_device[0] != _device) {
-		ATA_ERROR("an ASIO duplex stream must use the same device for input and output!");
-		return false;
-	}
-	char driverName[32];
-	ASIOError result = drivers.asioGetDriverName((int) _device, driverName, 32);
-	if (result != ASE_OK) {
-		ATA_ERROR("unable to get driver name (" << getAsioErrorString(result) << ").");
-		return false;
-	}
-	// Only load the driver once for duplex stream.
-	if (    _mode != audio::orchestra::mode_input
-	     || m_mode != audio::orchestra::mode_output) {
-		// The getDeviceInfo() function will not work when a stream is open
-		// because ASIO does not allow multiple devices to run at the same
-		// time.	Thus, we'll probe the system before opening a stream and
-		// save the results for use by getDeviceInfo().
-		this->saveDeviceInfo();
-		if (!drivers.loadDriver(driverName)) {
-			ATA_ERROR("unable to load driver (" << driverName << ").");
-			return false;
-		}
-		result = ASIOInit(&driverInfo);
-		if (result != ASE_OK) {
-			ATA_ERROR("error (" << getAsioErrorString(result) << ") initializing driver (" << driverName << ").");
-			return false;
-		}
-	}
-	// Check the device channel count.
-	long inputChannels, outputChannels;
-	result = ASIOGetChannels(&inputChannels, &outputChannels);
-	if (result != ASE_OK) {
-		drivers.removeCurrentDriver();
-		ATA_ERROR("error (" << getAsioErrorString(result) << ") getting channel count (" << driverName << ").");
-		return false;
-	}
-	if (    (    _mode == audio::orchestra::mode_output
-	          && (_channels+_firstChannel) > (uint32_t) outputChannels)
-	     || (    _mode == audio::orchestra::mode_input
-	          && (_channels+_firstChannel) > (uint32_t) inputChannels)) {
-		drivers.removeCurrentDriver();
-		ATA_ERROR("driver (" << driverName << ") does not support requested channel count (" << _channels << ") + offset (" << _firstChannel << ").");
-		return false;
-	}
-	m_nDeviceChannels[modeToIdTable(_mode)] = _channels;
-	m_nUserChannels[modeToIdTable(_mode)] = _channels;
-	m_channelOffset[modeToIdTable(_mode)] = _firstChannel;
-	// Verify the sample rate is supported.
-	result = ASIOCanSampleRate((ASIOSampleRate) _sampleRate);
-	if (result != ASE_OK) {
-		drivers.removeCurrentDriver();
-		ATA_ERROR("driver (" << driverName << ") does not support requested sample rate (" << _sampleRate << ").");
-		return false;
-	}
-	// Get the current sample rate
-	ASIOSampleRate currentRate;
-	result = ASIOGetSampleRate(&currentRate);
-	if (result != ASE_OK) {
-		drivers.removeCurrentDriver();
-		ATA_ERROR("driver (" << driverName << ") error getting sample rate.");
-		return false;
-	}
-	// Set the sample rate only if necessary
-	if (currentRate != _sampleRate) {
-		result = ASIOSetSampleRate((ASIOSampleRate) _sampleRate);
-		if (result != ASE_OK) {
-			drivers.removeCurrentDriver();
-			ATA_ERROR("driver (" << driverName << ") error setting sample rate (" << _sampleRate << ").");
-			return false;
-		}
-	}
-	// Determine the driver data type.
-	ASIOChannelInfo channelInfo;
-	channelInfo.channel = 0;
-	if (_mode == audio::orchestra::mode_output) {
-		channelInfo.isInput = false;
-	} else {
-		channelInfo.isInput = true;
-	}
-	result = ASIOGetChannelInfo(&channelInfo);
-	if (result != ASE_OK) {
-		drivers.removeCurrentDriver();
-		ATA_ERROR("driver (" << driverName << ") error (" << getAsioErrorString(result) << ") getting data format.");
-		return false;
-	}
-	// Assuming WINDOWS host is always little-endian.
-	m_doByteSwap[modeToIdTable(_mode)] = false;
-	m_userFormat = _format;
-	m_deviceFormat[modeToIdTable(_mode)] = 0;
-	if (    channelInfo.type == ASIOSTInt16MSB
-	     || channelInfo.type == ASIOSTInt16LSB) {
-		m_deviceFormat[modeToIdTable(_mode)] = RTAUDIO_SINT16;
-		if (channelInfo.type == ASIOSTInt16MSB) {
-			m_doByteSwap[modeToIdTable(_mode)] = true;
-		}
-	} else if (    channelInfo.type == ASIOSTInt32MSB
-	            || channelInfo.type == ASIOSTInt32LSB) {
-		m_deviceFormat[modeToIdTable(_mode)] = RTAUDIO_SINT32;
-		if (channelInfo.type == ASIOSTInt32MSB) {
-			m_doByteSwap[modeToIdTable(_mode)] = true;
-		}
-	} else if (    channelInfo.type == ASIOSTFloat32MSB
-	            || channelInfo.type == ASIOSTFloat32LSB) {
-		m_deviceFormat[modeToIdTable(_mode)] = RTAUDIO_FLOAT32;
-		if (channelInfo.type == ASIOSTFloat32MSB) {
-			m_doByteSwap[modeToIdTable(_mode)] = true;
-		}
-	} else if (    channelInfo.type == ASIOSTFloat64MSB
-	            || channelInfo.type == ASIOSTFloat64LSB) {
-		m_deviceFormat[modeToIdTable(_mode)] = RTAUDIO_FLOAT64;
-		if (channelInfo.type == ASIOSTFloat64MSB) {
-			m_doByteSwap[modeToIdTable(_mode)] = true;
-		}
-	} else if (    channelInfo.type == ASIOSTInt24MSB
-	            || channelInfo.type == ASIOSTInt24LSB) {
-		m_deviceFormat[modeToIdTable(_mode)] = RTAUDIO_SINT24;
-		if (channelInfo.type == ASIOSTInt24MSB) {
-			m_doByteSwap[modeToIdTable(_mode)] = true;
-		}
-	}
-	if (m_deviceFormat[modeToIdTable(_mode)] == 0) {
-		drivers.removeCurrentDriver();
-		ATA_ERROR("driver (" << driverName << ") data format not supported by RtAudio.");
-		return false;
-	}
-	// Set the buffer size.	For a duplex stream, this will end up
-	// setting the buffer size based on the input constraints, which
-	// should be ok.
-	long minSize, maxSize, preferSize, granularity;
-	result = ASIOGetBufferSize(&minSize, &maxSize, &preferSize, &granularity);
-	if (result != ASE_OK) {
-		drivers.removeCurrentDriver();
-		ATA_ERROR("driver (" << driverName << ") error (" << getAsioErrorString(result) << ") getting buffer size.");
-		return false;
-	}
-	if (*_bufferSize < (uint32_t) minSize) {
-		*_bufferSize = (uint32_t) minSize;
-	} else if (*_bufferSize > (uint32_t) maxSize) {
-		*_bufferSize = (uint32_t) maxSize;
-	} else if (granularity == -1) {
-		// Make sure bufferSize is a power of two.
-		int32_t log2_of_min_size = 0;
-		int32_t log2_of_max_size = 0;
-		for (uint32_t i = 0; i < sizeof(long) * 8; i++) {
-			if (minSize & ((long)1 << i)) {
-				log2_of_min_size = i;
-			}
-			if (maxSize & ((long)1 << i)) {
-				log2_of_max_size = i;
-			}
-		}
-		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 = etk::abs((long)*_bufferSize - ((long)1 << i));
-			if (current_delta < min_delta) {
-				min_delta = current_delta;
-				min_delta_num = i;
-			}
-		}
-		*_bufferSize = ((uint32_t)1 << min_delta_num);
-		if (*_bufferSize < (uint32_t) {
-			minSize) *_bufferSize = (uint32_t) minSize;
-		} else if (*_bufferSize > (uint32_t) maxSize) {
-			*_bufferSize = (uint32_t) maxSize;
-		}
-	} else if (granularity != 0) {
-		// Set to an even multiple of granularity, rounding up.
-		*_bufferSize = (*_bufferSize + granularity-1) / granularity * granularity;
-	}
-	if (    _mode == audio::orchestra::mode_input
-	     && m_mode == audio::orchestra::mode_output
-	     && m_bufferSize != *_bufferSize) {
-		drivers.removeCurrentDriver();
-		ATA_ERROR("input/output buffersize discrepancy!");
-		return false;
-	}
-	m_bufferSize = *_bufferSize;
-	m_nBuffers = 2;
-	// ASIO always uses non-interleaved buffers.
-	m_deviceInterleaved[modeToIdTable(_mode)] = false;
-	m_private->bufferInfos = 0;
-	// Create a manual-reset event.
-	m_private->condition = CreateEvent(null,  // no security
-	                                   TRUE,  // manual-reset
-	                                   FALSE, // non-signaled initially
-	                                   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.
-	long inputLatency, outputLatency;
-	if (    _mode == audio::orchestra::mode_input
-	     && m_mode == audio::orchestra::mode_output) {
-		ASIODisposeBuffers();
-		if (m_private->bufferInfos == null) {
-			free(m_private->bufferInfos);
-			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 == null) {
-		ATA_ERROR("error allocating bufferInfo memory for driver (" << driverName << ").");
-		goto error;
-	}
-	ASIOBufferInfo *infos;
-	infos = m_private->bufferInfos;
-	for (i=0; i<m_nDeviceChannels[0]; i++, infos++) {
-		infos->isInput = ASIOFalse;
-		infos->channelNum = i + m_channelOffset[0];
-		infos->buffers[0] = infos->buffers[1] = 0;
-	}
-	for (i=0; i<m_nDeviceChannels[1]; i++, infos++) {
-		infos->isInput = ASIOTrue;
-		infos->channelNum = i + m_channelOffset[1];
-		infos->buffers[0] = infos->buffers[1] = 0;
-	}
-	// Set up the ASIO callback structure and create the ASIO data buffers.
-	asioCallbacks.bufferSwitch = &bufferSwitch;
-	asioCallbacks.sampleRateDidChange = &sampleRateChanged;
-	asioCallbacks.asioMessage = &asioMessages;
-	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.");
-		goto error;
-	}
-	buffersAllocated = true;
-	// Set flags for buffer conversion.
-	m_doConvertBuffer[modeToIdTable(_mode)] = false;
-	if (m_userFormat != m_deviceFormat[modeToIdTable(_mode)]) {
-		m_doConvertBuffer[modeToIdTable(_mode)] = true;
-	}
-	if (    m_deviceInterleaved[modeToIdTable(_mode)] == false
-	     && m_nUserChannels[modeToIdTable(_mode)] > 1) {
-		m_doConvertBuffer[modeToIdTable(_mode)] = true;
-	}
-	// Allocate necessary internal buffers
-	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)] == null) {
-		ATA_ERROR("error allocating user buffer memory.");
-		goto error;
-	}
-	if (m_doConvertBuffer[modeToIdTable(_mode)]) {
-		bool makeBuffer = true;
-		bufferBytes = m_nDeviceChannels[modeToIdTable(_mode)] * audio::getFormatBytes(m_deviceFormat[modeToIdTable(_mode)]);
-		if (_mode == audio::orchestra::mode_input) {
-			if (m_mode == audio::orchestra::mode_output && m_deviceBuffer) {
-				uint64_t bytesOut = m_nDeviceChannels[0] * audio::getFormatBytes(m_deviceFormat[0]);
-				if (bufferBytes <= bytesOut) {
-					makeBuffer = false;
-				}
-			}
-		}
-		if (makeBuffer) {
-			bufferBytes *= *_bufferSize;
-			if (m_deviceBuffer) {
-				free(m_deviceBuffer);
-				m_deviceBuffer = null;
-			}
-			m_deviceBuffer = (char *) calloc(bufferBytes, 1);
-			if (m_deviceBuffer == null) {
-				ATA_ERROR("error allocating device buffer memory.");
-				goto error;
-			}
-		}
-	}
-	m_sampleRate = _sampleRate;
-	m_device[modeToIdTable(_mode)] = _device;
-	m_state = audio::orchestra::state::stopped;
-	if (    _mode == audio::orchestra::mode_output
-	     && _mode == audio::orchestra::mode_input) {
-		// We had already set up an output stream.
-		m_mode = audio::orchestra::mode_duplex;
-	} else {
-		m_mode = _mode;
-	}
-	// Determine device latencies
-	result = ASIOGetLatencies(&inputLatency, &outputLatency);
-	if (result != ASE_OK) {
-		ATA_ERROR("driver (" << driverName << ") error (" << getAsioErrorString(result) << ") getting latency.");
-	} else {
-		m_latency[0] = outputLatency;
-		m_latency[1] = inputLatency;
-	}
-	// Setup the buffer conversion information structure.	We don't use
-	// buffers to do channel offsets, so we override that parameter
-	// here.
-	if (m_doConvertBuffer[modeToIdTable(_mode)]) {
-		setConvertInfo(_mode, 0);
-	}
-	return true;
-error:
-	if (buffersAllocated) {
-		ASIODisposeBuffers();
-	}
-	drivers.removeCurrentDriver();
-	CloseHandle(m_private->condition);
-	if (m_private->bufferInfos != null) {
-		free(m_private->bufferInfos);
-		m_private->bufferInfos = null;
-	}
-	for (int32_t i=0; i<2; i++) {
-		if (m_userBuffer[i]) {
-			free(m_userBuffer[i]);
-			m_userBuffer[i] = 0;
-		}
-	}
-	if (m_deviceBuffer) {
-		free(m_deviceBuffer);
-		m_deviceBuffer = 0;
-	}
-	return false;
-}
-
-enum audio::orchestra::error audio::orchestra::api::Asio::closeStream() {
-	if (m_state == audio::orchestra::state::closed) {
-		ATA_ERROR("no open stream to close!");
-		return audio::orchestra::error_warning;
-	}
-	if (m_state == audio::orchestra::state::running) {
-		m_state = audio::orchestra::state::stopped;
-		ASIOStop();
-	}
-	ASIODisposeBuffers();
-	drivers.removeCurrentDriver();
-	CloseHandle(m_private->condition);
-	if (m_private->bufferInfos) {
-		free(m_private->bufferInfos);
-	}
-	for (int32_t i=0; i<2; i++) {
-		if (m_userBuffer[i]) {
-			free(m_userBuffer[i]);
-			m_userBuffer[i] = 0;
-		}
-	}
-	if (m_deviceBuffer) {
-		free(m_deviceBuffer);
-		m_deviceBuffer = 0;
-	}
-	m_mode = audio::orchestra::mode_unknow;
-	m_state = audio::orchestra::state::closed;
-	return audio::orchestra::error_none;
-}
-
-bool stopThreadCalled = false;
-
-enum audio::orchestra::error audio::orchestra::api::Asio::startStream() {
-	// TODO : Check return ...
-	audio::orchestra::Api::startStream();
-	if (verifyStream() != audio::orchestra::error_none) {
-		return audio::orchestra::error_fail;
-	}
-	if (m_state == audio::orchestra::state::running) {
-		ATA_ERROR("the stream is already running!");
-		return audio::orchestra::error_warning;
-	}
-	ASIOError result = ASIOStart();
-	if (result != ASE_OK) {
-		ATA_ERROR("error (" << getAsioErrorString(result) << ") starting device.");
-		goto unlock;
-	}
-	m_private->drainCounter = 0;
-	m_private->internalDrain = false;
-	ResetEvent(m_private->condition);
-	m_state = audio::orchestra::state::running;
-	asioXRun = false;
-unlock:
-	stopThreadCalled = false;
-	if (result == ASE_OK) {
-		return audio::orchestra::error_none;
-	}
-	return audio::orchestra::error_systemError;
-}
-
-enum audio::orchestra::error audio::orchestra::api::Asio::stopStream() {
-	if (verifyStream() != audio::orchestra::error_none) {
-		return audio::orchestra::error_fail;
-	}
-	if (m_state == audio::orchestra::state::stopped) {
-		ATA_ERROR("the stream is already stopped!");
-		return audio::orchestra::error_warning;
-	}
-	if (m_mode == audio::orchestra::mode_output || m_mode == audio::orchestra::mode_duplex) {
-		if (m_private->drainCounter == 0) {
-			m_private->drainCounter = 2;
-			WaitForSingleObject(m_private->condition, INFINITE);	// block until signaled
-		}
-	}
-	m_state = audio::orchestra::state::stopped;
-	ASIOError result = ASIOStop();
-	if (result != ASE_OK) {
-		ATA_ERROR("error (" << getAsioErrorString(result) << ") stopping device.");
-	}
-	if (result == ASE_OK) {
-		return audio::orchestra::error_none;
-	}
-	return audio::orchestra::error_systemError;
-}
-
-enum audio::orchestra::error audio::orchestra::api::Asio::abortStream() {
-	if (verifyStream() != audio::orchestra::error_none) {
-		return audio::orchestra::error_fail;
-	}
-	if (m_state == audio::orchestra::state::stopped) {
-		ATA_ERROR("the stream is already stopped!");
-		error(audio::orchestra::error_warning);
-		return;
-	}
-
-	// The following lines were commented-out because some behavior was
-	// noted where the device buffers need to be zeroed to avoid
-	// continuing sound, even when the device buffers are completely
-	// disposed.	So now, calling abort is the same as calling stop.
-	// handle->drainCounter = 2;
-	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 ASIOStop()
-// function will return.
-static unsigned __stdcall asioStopStream(void *_ptr) {
-	CallbackInfo* info = (CallbackInfo*)_ptr;
-	RtApiAsio* object = (RtApiAsio*)info->object;
-	object->stopStream();
-	_endthreadex(0);
-	return 0;
-}
-
-bool audio::orchestra::api::Asio::callbackEvent(long bufferIndex) {
-	if (    m_state == audio::orchestra::state::stopped
-	     || m_state == audio::orchestra::state::stopping) {
-		return true;
-	}
-	if (m_state == audio::orchestra::state::closed) {
-		ATA_ERROR("the stream is closed ... this shouldn't happen!");
-		return false;
-	}
-	CallbackInfo *info = (CallbackInfo *) &m_callbackInfo;
-	// Check if we were draining the stream and signal if finished.
-	if (m_private->drainCounter > 3) {
-		m_state = audio::orchestra::state::stopping;
-		if (m_private->internalDrain == false) {
-			SetEvent(m_private->condition);
-		} else { // spawn a thread to stop the stream
-			unsigned threadId;
-			m_callbackInfo.thread = _beginthreadex(null,
-			                                       0,
-			                                       &asioStopStream,
-			                                       &m_callbackInfo,
-			                                       0,
-			                                       &threadId);
-		}
-		return true;
-	}
-	// Invoke user callback to get fresh output data UNLESS we are
-	// draining stream.
-	if (m_private->drainCounter == 0) {
-		audio::Time streamTime = getStreamTime();
-		etk::Vector<enum audio::orchestra::status status;
-		if (m_mode != audio::orchestra::mode_input && asioXRun == true) {
-			status.pushBack(audio::orchestra::status::underflow);
-			asioXRun = false;
-		}
-		if (m_mode != audio::orchestra::mode_output && asioXRun == true) {
-			status.pushBack(audio::orchestra::status::underflow;
-			asioXRun = false;
-		}
-		int32_t cbReturnValue = info->callback(m_userBuffer[1],
-		                                       streamTime,
-		                                       m_userBuffer[0],
-		                                       streamTime,
-		                                       m_bufferSize,
-		                                       status);
-		if (cbReturnValue == 2) {
-			m_state = audio::orchestra::state::stopping;
-			m_private->drainCounter = 2;
-			unsigned threadId;
-			m_callbackInfo.thread = _beginthreadex(null,
-			                                              0,
-			                                              &asioStopStream,
-			                                              &m_callbackInfo,
-			                                              0,
-			                                              &threadId);
-			return true;
-		} else if (cbReturnValue == 1) {
-			m_private->drainCounter = 1;
-			m_private->internalDrain = true;
-		}
-	}
-	uint32_t nChannels, bufferBytes, i, j;
-	nChannels = m_nDeviceChannels[0] + m_nDeviceChannels[1];
-	if (    m_mode == audio::orchestra::mode_output
-	     || m_mode == audio::orchestra::mode_duplex) {
-		bufferBytes = m_bufferSize * audio::getFormatBytes(m_deviceFormat[0]);
-		if (m_private->drainCounter > 1) { // write zeros to the output stream
-			for (i=0, j=0; i<nChannels; i++) {
-				if (m_private->bufferInfos[i].isInput != ASIOTrue) {
-					memset(m_private->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes);
-				}
-			}
-		} else if (m_doConvertBuffer[0]) {
-			convertBuffer(m_deviceBuffer, m_userBuffer[0], m_convertInfo[0]);
-			if (m_doByteSwap[0]) {
-				byteSwapBuffer(m_deviceBuffer,
-				               m_bufferSize * m_nDeviceChannels[0],
-				               m_deviceFormat[0]);
-			}
-			for (i=0, j=0; i<nChannels; i++) {
-				if (m_private->bufferInfos[i].isInput != ASIOTrue) {
-					memcpy(m_private->bufferInfos[i].buffers[bufferIndex],
-					       &m_deviceBuffer[j++*bufferBytes],
-					       bufferBytes);
-				}
-			}
-		} else {
-			if (m_doByteSwap[0]) {
-				byteSwapBuffer(m_userBuffer[0],
-				               m_bufferSize * m_nUserChannels[0],
-				               m_userFormat);
-			}
-			for (i=0, j=0; i<nChannels; i++) {
-				if (m_private->bufferInfos[i].isInput != ASIOTrue) {
-					memcpy(m_private->bufferInfos[i].buffers[bufferIndex],
-					       &m_userBuffer[0][bufferBytes*j++],
-					       bufferBytes);
-				}
-			}
-		}
-		if (m_private->drainCounter) {
-			m_private->drainCounter++;
-			goto unlock;
-		}
-	}
-	if (    m_mode == audio::orchestra::mode_input
-	     || m_mode == audio::orchestra::mode_duplex) {
-		bufferBytes = m_bufferSize * audio::getFormatBytes(m_deviceFormat[1]);
-		if (m_doConvertBuffer[1]) {
-			// Always interleave ASIO input data.
-			for (i=0, j=0; i<nChannels; i++) {
-				if (m_private->bufferInfos[i].isInput == ASIOTrue) {
-					memcpy(&m_deviceBuffer[j++*bufferBytes],
-					       m_private->bufferInfos[i].buffers[bufferIndex],
-					       bufferBytes);
-				}
-			}
-			if (m_doByteSwap[1]) {
-				byteSwapBuffer(m_deviceBuffer,
-				               m_bufferSize * m_nDeviceChannels[1],
-				               m_deviceFormat[1]);
-			}
-			convertBuffer(m_userBuffer[1],
-			              m_deviceBuffer,
-			              m_convertInfo[1]);
-		} else {
-			for (i=0, j=0; i<nChannels; i++) {
-				if (m_private->bufferInfos[i].isInput == ASIOTrue) {
-					memcpy(&m_userBuffer[1][bufferBytes*j++],
-					       m_private->bufferInfos[i].buffers[bufferIndex],
-					       bufferBytes);
-				}
-			}
-			if (m_doByteSwap[1]) {
-				byteSwapBuffer(m_userBuffer[1],
-				               m_bufferSize * m_nUserChannels[1],
-				               m_userFormat);
-			}
-		}
-	}
-unlock:
-	// The following call was suggested by Malte Clasen.	While the API
-	// documentation indicates it should not be required, some device
-	// drivers apparently do not function correctly without it.
-	ASIOOutputReady();
-	audio::orchestra::Api::tickStreamTime();
-	return true;
-}
-
-static void sampleRateChanged(ASIOSampleRate _sRate) {
-	// The ASIO documentation says that this usually only happens during
-	// external sync.	Audio processing is not stopped by the driver,
-	// actual sample rate might not have even changed, maybe only the
-	// sample rate status of an AES/EBU or S/PDIF digital input at the
-	// audio device.
-	RtApi* object = (RtApi*)asioCallbackInfo->object;
-	enum audio::orchestra::error ret = object->stopStream()
-	if (ret != audio::orchestra::error_none) {
-		ATA_ERROR("error stop stream!");
-	} else {
-		ATA_ERROR("driver reports sample rate changed to " << _sRate << " ... stream stopped!!!");
-	}
-}
-
-static long asioMessages(long _selector, long _value, void* _message, double* _opt) {
-	long ret = 0;
-	switch(_selector) {
-		case kAsioSelectorSupported:
-			if (    _value == kAsioResetRequest
-			     || _value == kAsioEngineVersion
-			     || _value == kAsioResyncRequest
-			     || _value == kAsioLatenciesChanged
-			     // The following three were added for ASIO 2.0, you don't
-			     // necessarily have to support them.
-			     || _value == kAsioSupportsTimeInfo
-			     || _value == kAsioSupportsTimeCode
-			     || _value == kAsioSupportsInputMonitor) {
-				ret = 1L;
-			}
-			break;
-		case kAsioResetRequest:
-			// Defer the task and perform the reset of the driver during the
-			// next "safe" situation.	You cannot reset the driver right now,
-			// as this code is called from the driver.	Reset the driver is
-			// done by completely destruct is. I.e. ASIOStop(),
-			// ASIODisposeBuffers(), Destruction Afterwards you initialize the
-			// driver again.
-			ATA_ERROR("driver reset requested!!!");
-			ret = 1L;
-			break;
-		case kAsioResyncRequest:
-			// This informs the application that the driver encountered some
-			// non-fatal data loss.	It is used for synchronization purposes
-			// of different media.	Added mainly to work around the Win16Mutex
-			// problems in Windows 95/98 with the Windows Multimedia system,
-			// which could lose data because the Mutex was held too long by
-			// another thread.	However a driver can issue it in other
-			// situations, too.
-			// ATA_ERROR("driver resync requested!!!");
-			asioXRun = true;
-			ret = 1L;
-			break;
-		case kAsioLatenciesChanged:
-			// This will inform the host application that the drivers were
-			// latencies changed.	Beware, it this does not mean that the
-			// buffer sizes have changed!	You might need to update internal
-			// delay data.
-			ATA_ERROR("driver latency may have changed!!!");
-			ret = 1L;
-			break;
-		case kAsioEngineVersion:
-			// Return the supported ASIO version of the host application.	If
-			// a host application does not implement this selector, ASIO 1.0
-			// is assumed by the driver.
-			ret = 2L;
-			break;
-		case kAsioSupportsTimeInfo:
-			// Informs the driver whether the
-			// asioCallbacks.bufferSwitchTimeInfo() callback is supported.
-			// For compatibility with ASIO 1.0 drivers the host application
-			// should always support the "old" bufferSwitch method, too.
-			ret = 0;
-			break;
-		case kAsioSupportsTimeCode:
-			// Informs the driver whether application is interested in time
-			// code info.	If an application does not need to know about time
-			// code, the driver has less work to do.
-			ret = 0;
-			break;
-	}
-	return ret;
-}
-
-static const char* getAsioErrorString(ASIOError _result) {
-	struct Messages {
-		ASIOError value;
-		const char*message;
-	};
-	static const Messages m[] = {
-		{ ASE_NotPresent, "Hardware input or output is not present or available." },
-		{ ASE_HWMalfunction, "Hardware is malfunctioning." },
-		{ ASE_InvalidParameter, "Invalid input parameter." },
-		{ ASE_InvalidMode, "Invalid mode." },
-		{ ASE_SPNotAdvancing, "Sample position not advancing." },
-		{ ASE_NoClock, "Sample clock or rate cannot be determined or is not present." },
-		{ ASE_NoMemory, "Not enough memory to complete the request." }
-	};
-	for (uint32_t i = 0; i < sizeof(m)/sizeof(m[0]); ++i) {
-		if (m[i].value == result) {
-			return m[i].message;
-		}
-	}
-	return "Unknown error.";
-}
-
-#endif

audio/orchestra/api/Asio.hpp

@@ -1,53 +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
-#ifdef ORCHESTRA_BUILD_ASIO
-
-namespace audio {
-	namespace orchestra {
-		namespace api {
-			class AsioPrivate:
-			class Asio: public audio::orchestra::Api {
-				public:
-					static ememory::SharedPtr<audio::orchestra::Api> create();
-				public:
-					Asio();
-					virtual ~Asio();
-					const etk::String& getCurrentApi() {
-						return audio::orchestra::typeAsio;
-					}
-					uint32_t getDeviceCount();
-					audio::orchestra::DeviceInfo getDeviceInfo(uint32_t _device);
-					enum audio::orchestra::error closeStream();
-					enum audio::orchestra::error startStream();
-					enum audio::orchestra::error stopStream();
-					enum audio::orchestra::error abortStream();
-					long getStreamLatency();
-					// This function is intended for internal use only.	It must be
-					// public because it is called by the internal callback handler,
-					// which is not a member of RtAudio.	External use of this function
-					// will most likely produce highly undesireable results!
-					bool callbackEvent(long _bufferIndex);
-				private:
-					ememory::SharedPtr<AsioPrivate> m_private;
-					etk::Vector<audio::orchestra::DeviceInfo> m_devices;
-					void saveDeviceInfo();
-					bool m_coInitialized;
-					bool open(uint32_t _device,
-					          audio::orchestra::mode _mode,
-					          uint32_t _channels,
-					          uint32_t _firstChannel,
-					          uint32_t _sampleRate,
-					          audio::format _format,
-					          uint32_t *_bufferSize,
-					          const audio::orchestra::StreamOptions& _options);
-			};
-		}
-	}
-}
-
-#endif

audio/orchestra/api/Core.hpp

@@ -1,68 +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
-#ifdef ORCHESTRA_BUILD_MACOSX_CORE
-
-#include <CoreAudio/AudioHardware.h>
-
-
-namespace audio {
-	namespace orchestra {
-		namespace api {
-			class CorePrivate;
-			class Core: public audio::orchestra::Api {
-				public:
-					static ememory::SharedPtr<audio::orchestra::Api> create();
-				public:
-					Core();
-					virtual ~Core();
-					const etk::String& getCurrentApi() {
-						return audio::orchestra::typeCoreOSX;
-					}
-					uint32_t getDeviceCount();
-					audio::orchestra::DeviceInfo getDeviceInfo(uint32_t _device);
-					uint32_t getDefaultOutputDevice();
-					uint32_t getDefaultInputDevice();
-					enum audio::orchestra::error closeStream();
-					enum audio::orchestra::error startStream();
-					enum audio::orchestra::error stopStream();
-					enum audio::orchestra::error abortStream();
-					long getStreamLatency();
-					bool callbackEvent(AudioDeviceID _deviceId,
-					                   const AudioBufferList *_inBufferList,
-					                   const audio::Time& _inTime,
-					                   const AudioBufferList *_outBufferList,
-					                   const audio::Time& _outTime);
-					static OSStatus callbackEvent(AudioDeviceID _inDevice,
-					                              const AudioTimeStamp* _inNow,
-					                              const AudioBufferList* _inInputData,
-					                              const AudioTimeStamp* _inInputTime,
-					                              AudioBufferList* _outOutputData,
-					                              const AudioTimeStamp* _inOutputTime,
-					                              void* _infoPointer);
-					static void coreStopStream(void *_userData);
-				private:
-					ememory::SharedPtr<CorePrivate> m_private;
-					bool open(uint32_t _device,
-					          audio::orchestra::mode _mode,
-					          uint32_t _channels,
-					          uint32_t _firstChannel,
-					          uint32_t _sampleRate,
-					          audio::format _format,
-					          uint32_t *_bufferSize,
-					          const audio::orchestra::StreamOptions& _options);
-					static const char* getErrorCode(OSStatus _code);
-					static OSStatus xrunListener(AudioObjectID _inDevice,
-					                             uint32_t _nAddresses,
-					                             const AudioObjectPropertyAddress _properties[],
-					                             void* _userData);
-			};
-		}
-	}
-}
-
-#endif

audio/orchestra/api/CoreIos.hpp

@@ -1,58 +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
-#ifdef ORCHESTRA_BUILD_IOS_CORE
-
-namespace audio {
-	namespace orchestra {
-		namespace api {
-			class CoreIosPrivate;
-			class CoreIos: public audio::orchestra::Api {
-				public:
-					static ememory::SharedPtr<audio::orchestra::Api> create();
-				public:
-					CoreIos();
-					virtual ~CoreIos();
-					const etk::String& getCurrentApi() {
-						return audio::orchestra::typeCoreIOS;
-					}
-					uint32_t getDeviceCount();
-					audio::orchestra::DeviceInfo getDeviceInfo(uint32_t _device);
-					enum audio::orchestra::error closeStream();
-					enum audio::orchestra::error startStream();
-					enum audio::orchestra::error stopStream();
-					enum audio::orchestra::error abortStream();
-					// This function is intended for internal use only.	It must be
-					// public because it is called by the internal callback handler,
-					// which is not a member of RtAudio.	External use of this function
-					// will most likely produce highly undesireable results!
-					void callbackEvent();
-				private:
-					etk::Vector<audio::orchestra::DeviceInfo> m_devices;
-					void saveDeviceInfo();
-					bool open(uint32_t _device,
-					          audio::orchestra::mode _mode,
-					          uint32_t _channels,
-					          uint32_t _firstChannel,
-					          uint32_t _sampleRate,
-					          audio::format _format,
-					          uint32_t *_bufferSize,
-					          const audio::orchestra::StreamOptions& _options);
-				public:
-					void callBackEvent(void* _data,
-					                   int32_t _nbChunk,
-					                   const audio::Time& _time);
-				public:
-					ememory::SharedPtr<CoreIosPrivate> m_private;
-					uint32_t getDefaultInputDevice();
-					uint32_t getDefaultOutputDevice();
-			};
-		}
-	}
-}
-
-#endif

audio/orchestra/api/CoreIos.mm

@@ -1,336 +0,0 @@
-/** @file
- * @author Edouard DUPIN 
- * @copyright 2011, Edouard DUPIN, all right reserved
- * @license APACHE v2.0 (see license file)
- * @fork from RTAudio
- */
-
-#ifdef ORCHESTRA_BUILD_IOS_CORE
-
-#import <Foundation/Foundation.h>
-#import <AudioToolbox/AudioToolbox.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>(ETK_NEW(audio::orchestra::api::CoreIos));
-}
-
-#define kOutputBus 0
-#define kInputBus 1
-
-namespace audio {
-	namespace orchestra {
-		namespace api {
-			class CoreIosPrivate {
-				public:
-					AudioComponentInstance audioUnit;
-			};
-		}
-	}
-}
-
-
-
-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.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.nativeFormats.push_back(audio::format_int16);
-	tmp.isDefault = true;
-	tmp.isCorrect = true;
-	m_devices.push_back(tmp);
-	ATA_INFO("Create CoreIOs interface (end)");
-}
-
-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() {
-	//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() {
-	ATA_INFO("Close Stream");
-	// Can not close the stream now...
-	return audio::orchestra::error_none;
-}
-
-enum audio::orchestra::error audio::orchestra::api::CoreIos::startStream() {
-	ATA_INFO("Start Stream");
-	// TODO : Check return ...
-	audio::orchestra::Api::startStream();
-	OSStatus status = AudioOutputUnitStart(m_private->audioUnit);
-	// Can not close the stream now...
-	return audio::orchestra::error_none;
-}
-
-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() {
-	ATA_INFO("Abort Stream");
-	OSStatus status = AudioOutputUnitStop(m_private->audioUnit);
-	// Can not close the stream now...
-	return audio::orchestra::error_none;
-}
-
-void audio::orchestra::api::CoreIos::callBackEvent(void* _data,
-                                                   int32_t _nbChunk,
-                                                   const audio::Time& _time) {
-	int32_t doStopStream = 0;
-	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,
-		                          null,
-		                          audio::Time(),
-		                          _nbChunk,
-		                          status);
-	}
-	if (doStopStream == 2) {
-		abortStream();
-		return;
-	}
-	audio::orchestra::Api::tickStreamTime();
-}
-
-
-static OSStatus playbackCallback(void *_userData,
-                                 AudioUnitRenderActionFlags* _ioActionFlags,
-                                 const AudioTimeStamp* _inTime,
-                                 uint32_t _inBusNumber,
-                                 uint32_t _inNumberFrames,
-                                 AudioBufferList* _ioData) {
-	if (_userData == null) {
-		ATA_ERROR("callback event ... null pointer");
-		return -1;
-	}
-	audio::Time tmpTimeime;
-	if (_inTime != null) {
-		tmpTimeime = audio::Time(_inTime->mHostTime/1000000000LL, _inTime->mHostTime%1000000000LL);
-	}
-	audio::orchestra::api::CoreIos* myClass = static_cast<audio::orchestra::api::CoreIos*>(_userData);
-	// get all requested buffer :
-	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_INFO("request data size: " << numberFrame << " busNumber=" << _inBusNumber);
-		myClass->callBackEvent(buffer.mData, numberFrame, tmpTimeime);
-	}
-	return noErr;
-}
-
-
-bool audio::orchestra::api::CoreIos::open(uint32_t _device,
-                                          audio::orchestra::mode _mode,
-                                          uint32_t _channels,
-                                          uint32_t _firstChannel,
-                                          uint32_t _sampleRate,
-                                          audio::format _format,
-                                          uint32_t *_bufferSize,
-                                          const audio::orchestra::StreamOptions& _options) {
-	ATA_INFO("Probe : device=" << _device << " channels=" << _channels << " firstChannel=" << _firstChannel << " sampleRate=" << _sampleRate);
-	if (_mode != audio::orchestra::mode_output) {
-		ATA_ERROR("Can not start a device input or duplex for CoreIos ...");
-		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;
-	m_bufferSize = 8192;
-	m_sampleRate = _sampleRate;
-	m_doByteSwap[modeToIdTable(_mode)] = false; // for endienness ...
-	
-	// TODO : For now, we write it in hard ==> to be update later ...
-	m_deviceFormat[modeToIdTable(_mode)] = audio::format_int16;
-	m_nDeviceChannels[modeToIdTable(_mode)] = 2;
-	m_deviceInterleaved[modeToIdTable(_mode)] =	true;
-	
-	m_doConvertBuffer[modeToIdTable(_mode)] = false;
-	if (m_userFormat != m_deviceFormat[modeToIdTable(_mode)]) {
-		m_doConvertBuffer[modeToIdTable(_mode)] = true;
-	}
-	if (m_nUserChannels[modeToIdTable(_mode)] < m_nDeviceChannels[modeToIdTable(_mode)]) {
-		m_doConvertBuffer[modeToIdTable(_mode)] = true;
-	}
-	if (    m_deviceInterleaved[modeToIdTable(_mode)] == false
-	     && m_nUserChannels[modeToIdTable(_mode)] > 1) {
-		m_doConvertBuffer[modeToIdTable(_mode)] = true;
-	}
-	if (m_doConvertBuffer[modeToIdTable(_mode)] == true) {
-		// Allocate necessary internal buffers.
-		uint64_t bufferBytes = m_nUserChannels[modeToIdTable(_mode)] * m_bufferSize * audio::getFormatBytes(m_userFormat);
-		m_userBuffer[modeToIdTable(_mode)].resize(bufferBytes, 0);
-		if (m_userBuffer[modeToIdTable(_mode)].size() == 0) {
-			ATA_ERROR("error allocating user buffer memory.");
-		}
-		setConvertInfo(_mode, _firstChannel);
-	}
-	ATA_INFO("device format : " << m_deviceFormat[modeToIdTable(_mode)] << " user format : " << m_userFormat);
-	ATA_INFO("device channels : " << m_nDeviceChannels[modeToIdTable(_mode)] << " user channels : " << m_nUserChannels[modeToIdTable(_mode)]);
-	ATA_INFO("do convert buffer : " << m_doConvertBuffer[modeToIdTable(_mode)]);
-	if (ret == false) {
-		ATA_ERROR("Can not open device.");
-	}
-	
-	// Configure IOs interface:
-	OSStatus status;
-	
-	// Describe audio component
-	AudioComponentDescription desc;
-	desc.componentType = kAudioUnitType_Output;
-	desc.componentSubType = kAudioUnitSubType_RemoteIO;
-	desc.componentFlags = 0;
-	desc.componentFlagsMask = 0;
-	desc.componentManufacturer = kAudioUnitManufacturer_Apple;
-	
-	// Get component
-	AudioComponent inputComponent = AudioComponentFindNext(null, &desc);
-	
-	// Get audio units
-	status = AudioComponentInstanceNew(inputComponent, &m_private->audioUnit);
-	if (status != 0) {
-		ATA_ERROR("can not create an audio intance...");
-	}
-	
-	uint32_t flag = 1;
-	// Enable IO for playback
-	status = AudioUnitSetProperty(m_private->audioUnit, 
-	                              kAudioOutputUnitProperty_EnableIO, 
-	                              kAudioUnitScope_Output, 
-	                              kOutputBus,
-	                              &flag, 
-	                              sizeof(flag));
-	if (status != 0) {
-		ATA_ERROR("can not request audio autorisation...");
-	}
-	
-	// Describe format
-	AudioStreamBasicDescription audioFormat;
-	audioFormat.mSampleRate = 48000.00;
-	audioFormat.mFormatID = kAudioFormatLinearPCM;
-	audioFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
-	audioFormat.mFramesPerPacket = 1; //
-	audioFormat.mChannelsPerFrame = 2; // stereo
-	audioFormat.mBitsPerChannel = sizeof(short) * 8;
-	audioFormat.mBytesPerPacket = sizeof(short) * audioFormat.mChannelsPerFrame;
-	audioFormat.mBytesPerFrame = sizeof(short) * audioFormat.mChannelsPerFrame;
-	audioFormat.mReserved = 0;
-	// Apply format
-	status = AudioUnitSetProperty(m_private->audioUnit, 
-	                              kAudioUnitProperty_StreamFormat, 
-	                              kAudioUnitScope_Input, 
-	                              kOutputBus, 
-	                              &audioFormat, 
-	                              sizeof(audioFormat));
-	if (status != 0) {
-		ATA_ERROR("can not set stream properties...");
-	}
-	
-	
-	// Set output callback
-	AURenderCallbackStruct callbackStruct;
-	callbackStruct.inputProc = &playbackCallback;
-	callbackStruct.inputProcRefCon = this;
-	status = AudioUnitSetProperty(m_private->audioUnit, 
-	                              kAudioUnitProperty_SetRenderCallback, 
-	                              kAudioUnitScope_Global, 
-	                              kOutputBus,
-	                              &callbackStruct, 
-	                              sizeof(callbackStruct));
-	if (status != 0) {
-		ATA_ERROR("can not set Callback...");
-	}
-	
-	// Initialise
-	status = AudioUnitInitialize(m_private->audioUnit);
-	if (status != 0) {
-		ATA_ERROR("can not initialize...");
-	}
-	return ret;
-}
-
-#endif
-

audio/orchestra/api/Ds.hpp

@@ -1,54 +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
-#ifdef ORCHESTRA_BUILD_DS
-
-namespace audio {
-	namespace orchestra {
-		namespace api {
-			class DsPrivate;
-			class Ds: public audio::orchestra::Api {
-				public:
-					static ememory::SharedPtr<audio::orchestra::Api> create();
-				public:
-					Ds();
-					virtual ~Ds();
-					const etk::String& getCurrentApi() {
-						return audio::orchestra::typeDs;
-					}
-					uint32_t getDeviceCount();
-					audio::orchestra::DeviceInfo getDeviceInfo(uint32_t _device);
-					enum audio::orchestra::error closeStream();
-					enum audio::orchestra::error startStream();
-					enum audio::orchestra::error stopStream();
-					enum audio::orchestra::error abortStream();
-					long getStreamLatency();
-					// This function is intended for internal use only.	It must be
-					// public because it is called by the internal callback handler,
-					// which is not a member of RtAudio.	External use of this function
-					// will most likely produce highly undesireable results!
-					void callbackEvent();
-				private:
-					static void dsCallbackEvent(void *_userData);
-					ememory::SharedPtr<DsPrivate> m_private;
-					bool m_coInitialized;
-					bool m_buffersRolling;
-					long m_duplexPrerollBytes;
-					bool open(uint32_t _device,
-					          enum audio::orchestra::mode _mode,
-					          uint32_t _channels,
-					          uint32_t _firstChannel,
-					          uint32_t _sampleRate,
-					          enum audio::format _format,
-					          uint32_t *_bufferSize,
-					          const audio::orchestra::StreamOptions& _options);
-			};
-		}
-	}
-}
-
-#endif

audio/orchestra/api/Dummy.cpp

@@ -1,60 +0,0 @@
-/** @file
- * @author Edouard DUPIN 
- * @copyright 2011, Edouard DUPIN, all right reserved
- * @license APACHE v2.0 (see license file)
- * @fork from RTAudio
- */
-
-#if defined(ORCHESTRA_BUILD_DUMMY)
-#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>(ETK_NEW(audio::orchestra::api::Dummy));
-}
-
-
-audio::orchestra::api::Dummy::Dummy() {
-	ATA_WARNING("This class provides no functionality.");
-}
-
-uint32_t audio::orchestra::api::Dummy::getDeviceCount() {
-	return 0;
-}
-
-audio::orchestra::DeviceInfo audio::orchestra::api::Dummy::getDeviceInfo(uint32_t _device) {
-	(void)_device;
-	return audio::orchestra::DeviceInfo();
-}
-
-enum audio::orchestra::error audio::orchestra::api::Dummy::closeStream() {
-	return audio::orchestra::error_none;
-}
-
-enum audio::orchestra::error audio::orchestra::api::Dummy::startStream() {
-	// TODO : Check return ...
-	audio::orchestra::Api::startStream();
-	return audio::orchestra::error_none;
-}
-
-enum audio::orchestra::error audio::orchestra::api::Dummy::stopStream() {
-	return audio::orchestra::error_none;
-}
-
-enum audio::orchestra::error audio::orchestra::api::Dummy::abortStream() {
-	return audio::orchestra::error_none;
-}
-
-bool audio::orchestra::api::Dummy::open(uint32_t _device,
-                                        audio::orchestra::mode _mode,
-                                        uint32_t _channels,
-                                        uint32_t _firstChannel,
-                                        uint32_t _sampleRate,
-                                        audio::format _format,
-                                        uint32_t *_bufferSize,
-                                        const audio::orchestra::StreamOptions& _options) {
-	return false;
-}
-
-#endif
-

audio/orchestra/api/Dummy.hpp

@@ -1,44 +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
-
-#ifdef ORCHESTRA_BUILD_DUMMY
-
-#include <audio/orchestra/Interface.hpp>
-
-namespace audio {
-	namespace orchestra {
-		namespace api {
-			class Dummy: public audio::orchestra::Api {
-				public:
-					static ememory::SharedPtr<audio::orchestra::Api> create();
-				public:
-					Dummy();
-					const etk::String& getCurrentApi() {
-						return audio::orchestra::typeDummy;
-					}
-					uint32_t getDeviceCount();
-					audio::orchestra::DeviceInfo getDeviceInfo(uint32_t _device);
-					enum audio::orchestra::error closeStream();
-					enum audio::orchestra::error startStream();
-					enum audio::orchestra::error stopStream();
-					enum audio::orchestra::error abortStream();
-				private:
-					bool open(uint32_t _device,
-					          audio::orchestra::mode _mode,
-					          uint32_t _channels,
-					          uint32_t _firstChannel,
-					          uint32_t _sampleRate,
-					          audio::format _format,
-					          uint32_t *_bufferSize,
-					          const audio::orchestra::StreamOptions& _options);
-			};
-		}
-	}
-}
-
-#endif

audio/orchestra/api/Jack.cpp

@@ -1,712 +0,0 @@
-/** @file
- * @author Edouard DUPIN 
- * @copyright 2011, Edouard DUPIN, all right reserved
- * @license APACHE v2.0 (see license file)
- * @fork from RTAudio
- */
-
-// must run before :          
-#if defined(ORCHESTRA_BUILD_JACK)
-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>(ETK_NEW(audio::orchestra::api::Jack));
-}
-
-
-// JACK is a low-latency audio server, originally written for the
-// GNU/Linux operating system and now also ported to OS-X. It can
-// connect a number of different applications to an audio device, as
-// well as allowing them to share audio between themselves.
-//
-// When using JACK with RtAudio, "devices" refer to JACK clients that
-// have ports connected to the server. The JACK server is typically
-// started in a terminal as follows:
-//
-// .jackd -d alsa -d hw:0
-//
-// or through an interface program such as qjackctl. Many of the
-// parameters normally set for a stream are fixed by the JACK server
-// and can be specified when the JACK server is started. In
-// particular,
-//
-// jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4
-// jackd -r -d alsa -r 48000
-//
-// specifies a sample rate of 44100 Hz, a buffer size of 512 sample
-// frames, and number of buffers = 4. Once the server is running, it
-// is not possible to override these values. If the values are not
-// specified in the command-line, the JACK server uses default values.
-//
-// The JACK server does not have to be running when an instance of
-// audio::orchestra::Jack is created, though the function getDeviceCount() will
-// report 0 devices found until JACK has been started. When no
-// devices are available (i.e., the JACK server is not running), a
-// stream cannot be opened.
-
-#include <jack/jack.h>
-extern "C" {
-#include <stdio.h>
-}
-
-namespace audio {
-	namespace orchestra {
-		namespace api {
-			class JackPrivate {
-				public:
-					jack_client_t *client;
-					jack_port_t **ports[2];
-					etk::String deviceName[2];
-					bool xrun[2];
-					ethread::Semaphore m_semaphore;
-					int32_t drainCounter; // Tracks callback counts when draining
-					bool internalDrain; // Indicates if stop is initiated from callback or not.
-					
-					JackPrivate() :
-					  client(0),
-					  drainCounter(0),
-					  internalDrain(false) {
-						ports[0] = 0;
-						ports[1] = 0;
-						xrun[0] = false;
-						xrun[1] = false;
-				}
-			};
-		}
-	}
-}
-
-audio::orchestra::api::Jack::Jack() :
-  m_private(ETK_NEW(audio::orchestra::api::JackPrivate)) {
-	// Nothing to do here.
-}
-
-audio::orchestra::api::Jack::~Jack() {
-	if (m_state != audio::orchestra::state::closed) {
-		closeStream();
-	}
-}
-
-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 = null;
-	jack_client_t *client = jack_client_open("orchestraJackCount", options, status);
-	if (client == null) {
-		return 0;
-	}
-	const char **ports;
-	etk::String port, previousPort;
-	uint32_t nChannels = 0, nDevices = 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 != etk::String::npos) {
-				port = port.extract(0, iColon + 1);
-				if (port != previousPort) {
-					nDevices++;
-					previousPort = port;
-				}
-			}
-		} while (ports[++nChannels]);
-		free(ports);
-		ports = null;
-	}
-	jack_client_close(client);
-	return nDevices*2;
-}
-
-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 = null;
-	jack_client_t *client = jack_client_open("orchestraJackInfo", options, status);
-	if (client == null) {
-		ATA_ERROR("Jack server not found or connection error!");
-		// TODO : audio::orchestra::error_warning;
-		info.clear();
-		return info;
-	}
-	const char **ports;
-	etk::String port, previousPort;
-	uint32_t nPorts = 0, nDevices = 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) {
-		// Parse the port names up to the first colon (:).
-		size_t iColon = 0;
-		do {
-			port = (char *) ports[nPorts];
-			iColon = port.find(":");
-			if (iColon != etk::String::npos) {
-				port = port.extract(0, iColon);
-				if (port != previousPort) {
-					if (nDevices == deviceID) {
-						info.name = port;
-					}
-					nDevices++;
-					previousPort = port;
-				}
-			}
-		} while (ports[++nPorts]);
-		free(ports);
-	}
-	if (deviceID >= nDevices) {
-		jack_client_close(client);
-		ATA_ERROR("device ID is invalid!");
-		// TODO : audio::orchestra::error_invalidUse;
-		return info;
-	}
-	// Get the current jack server sample rate.
-	info.sampleRates.clear();
-	info.sampleRates.pushBack(jack_get_sample_rate(client));
-	if (info.input == true) {
-		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.pushBack(audio::channel_unknow);
-				iii++;
-			}
-			free(ports);
-		}
-	} else {
-		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.pushBack(audio::channel_unknow);
-				iii++;
-			}
-			free(ports);
-		}
-	}
-	if (info.channels.size() == 0) {
-		jack_client_close(client);
-		ATA_ERROR("error determining Jack input/output channels!");
-		// TODO : audio::orchestra::error_warning;
-		info.clear();
-		return info;
-	}
-	// Jack always uses 32-bit floats.
-	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;
-	}
-	jack_client_close(client);
-	info.isCorrect = true;
-	return info;
-}
-
-int32_t audio::orchestra::api::Jack::jackCallbackHandler(jack_nframes_t _nframes, void* _userData) {
-	ATA_VERBOSE("Jack callback: [BEGIN] " << uint64_t(_userData));
-	audio::orchestra::api::Jack* myClass = reinterpret_cast<audio::orchestra::api::Jack*>(_userData);
-	if (myClass->callbackEvent((uint64_t)_nframes) == false) {
-		ATA_VERBOSE("Jack callback: [END] 1");
-		return 1;
-	}
-	ATA_VERBOSE("Jack callback: [END] 0");
-	return 0;
-}
-
-
-void audio::orchestra::api::Jack::jackShutdown(void* _userData) {
-	audio::orchestra::api::Jack* myClass = reinterpret_cast<audio::orchestra::api::Jack*>(_userData);
-	// Check current stream state. If stopped, then we'll assume this
-	// was called as a result of a call to audio::orchestra::api::Jack::stopStream (the
-	// deactivation of a client handle causes this function to be called).
-	// If not, we'll assume the Jack server is shutting down or some
-	// other problem occurred and we should close the stream.
-	if (myClass->isStreamRunning() == false) {
-		return;
-	}
-	ETK_NEW(ethread::Thread, [=](){myClass->closeStream();});
-	ATA_ERROR("The Jack server is shutting down this client ... stream stopped and closed!!");
-}
-
-int32_t audio::orchestra::api::Jack::jackXrun(void* _userData) {
-	audio::orchestra::api::Jack* myClass = reinterpret_cast<audio::orchestra::api::Jack*>(_userData);
-	if (myClass->m_private->ports[0]) {
-		myClass->m_private->xrun[0] = true;
-	}
-	if (myClass->m_private->ports[1]) {
-		myClass->m_private->xrun[1] = true;
-	}
-	return 0;
-}
-
-bool audio::orchestra::api::Jack::open(uint32_t _device,
-                                       audio::orchestra::mode _mode,
-                                       uint32_t _channels,
-                                       uint32_t _firstChannel,
-                                       uint32_t _sampleRate,
-                                       audio::format _format,
-                                       uint32_t* _bufferSize,
-                                       const audio::orchestra::StreamOptions& _options) {
-	// Look for jack server and try to become a client (only do once per stream).
-	jack_client_t *client = 0;
-	if (    _mode == audio::orchestra::mode_output
-	     || (    _mode == audio::orchestra::mode_input
-	          && m_mode != audio::orchestra::mode_output)) {
-		jack_options_t jackoptions = (jack_options_t) (JackNoStartServer); //JackNullOption;
-		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);
-		}
-		if (client == 0) {
-			ATA_ERROR("Jack server not found or connection error!");
-			return false;
-		}
-	} else {
-		// The handle must have been created on an earlier pass.
-		client = m_private->client;
-	}
-	const char **ports;
-	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, 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 != etk::String::npos) {
-				port = port.extract(0, iColon);
-				if (port != previousPort) {
-					if (nDevices == deviceID) {
-						deviceName = port;
-					}
-					nDevices++;
-					previousPort = port;
-				}
-			}
-		} while (ports[++nPorts]);
-		free(ports);
-	}
-	if (_device >= nDevices) {
-		ATA_ERROR("device ID is invalid!");
-		return false;
-	}
-	// Count the available ports containing the client name as device
-	// channels.	Jack "input ports" equal RtAudio output channels.
-	uint32_t nChannels = 0;
-	uint64_t flag = JackPortIsInput;
-	if (_mode == audio::orchestra::mode_input) {
-		flag = JackPortIsOutput;
-	}
-	ports = jack_get_ports(client, deviceName.c_str(), null, flag);
-	if (ports) {
-		while (ports[ nChannels ]) {
-			nChannels++;
-		}
-		free(ports);
-	}
-	// Compare the jack ports for specified client to the requested number of channels.
-	if (nChannels < (_channels + _firstChannel)) {
-		ATA_ERROR("requested number of channels (" << _channels << ") + offset (" << _firstChannel << ") not found for specified device (" << _device << ":" << deviceName << ").");
-		return false;
-	}
-	// Check the jack server sample rate.
-	uint32_t jackRate = jack_get_sample_rate(client);
-	if (_sampleRate != jackRate) {
-		jack_client_close(client);
-		ATA_ERROR("the requested sample rate (" << _sampleRate << ") is different than the JACK server rate (" << jackRate << ").");
-		return false;
-	}
-	m_sampleRate = jackRate;
-	// Get the latency of the JACK port.
-	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);
-		// the range (usually the min and max are equal)
-		jack_latency_range_t latrange; latrange.min = latrange.max = 0;
-		// get the latency range
-		jack_port_get_latency_range(jack_port_by_name(client, ports[_firstChannel]), cbmode, &latrange);
-		// be optimistic, use the min!
-		m_latency[modeToIdTable(_mode)] = latrange.min;
-		//m_latency[modeToIdTable(_mode)] = jack_port_get_latency(jack_port_by_name(client, ports[ _firstChannel ]));
-	}
-	free(ports);
-	// The jack server always uses 32-bit floating-point data.
-	m_deviceFormat[modeToIdTable(_mode)] = audio::format_float;
-	m_userFormat = _format;
-	// Jack always uses non-interleaved buffers.
-	m_deviceInterleaved[modeToIdTable(_mode)] = false;
-	// Jack always provides host byte-ordered data.
-	m_doByteSwap[modeToIdTable(_mode)] = false;
-	// Get the buffer size.	The buffer size and number of buffers
-	// (periods) is set when the jack server is started.
-	m_bufferSize = (int) jack_get_buffer_size(client);
-	*_bufferSize = m_bufferSize;
-	m_nDeviceChannels[modeToIdTable(_mode)] = _channels;
-	m_nUserChannels[modeToIdTable(_mode)] = _channels;
-	// Set flags for buffer conversion.
-	m_doConvertBuffer[modeToIdTable(_mode)] = false;
-	if (m_userFormat != m_deviceFormat[modeToIdTable(_mode)]) {
-		m_doConvertBuffer[modeToIdTable(_mode)] = true;
-		ATA_CRITICAL("Can not update format ==> use RIVER lib for this ...");
-	}
-	if (    m_deviceInterleaved[modeToIdTable(_mode)] == false
-	     && m_nUserChannels[modeToIdTable(_mode)] > 1) {
-		ATA_ERROR("Reorder channel for the interleaving properties ...");
-		m_doConvertBuffer[modeToIdTable(_mode)] = true;
-	}
-	// Allocate our JackHandle structure for the stream.
-	m_private->client = client;
-	m_private->deviceName[modeToIdTable(_mode)] = deviceName;
-	// Allocate necessary internal buffers.
-	uint64_t bufferBytes;
-	bufferBytes = m_nUserChannels[modeToIdTable(_mode)] * *_bufferSize * audio::getFormatBytes(m_deviceFormat[modeToIdTable(_mode)]);
-	ATA_VERBOSE("allocate : nbChannel=" << m_nUserChannels[modeToIdTable(_mode)] << " bufferSize=" << *_bufferSize << " format=" << m_deviceFormat[modeToIdTable(_mode)] << "=" << audio::getFormatBytes(m_deviceFormat[modeToIdTable(_mode)]));
-	m_userBuffer[modeToIdTable(_mode)].resize(bufferBytes, 0);
-	if (m_userBuffer[modeToIdTable(_mode)].size() == 0) {
-		ATA_ERROR("error allocating user buffer memory.");
-		goto error;
-	}
-	if (m_doConvertBuffer[modeToIdTable(_mode)]) {
-		bool makeBuffer = true;
-		if (_mode == audio::orchestra::mode_output) {
-			bufferBytes = m_nDeviceChannels[0] * audio::getFormatBytes(m_deviceFormat[0]);
-		} else { // _mode == audio::orchestra::mode_input
-			bufferBytes = m_nDeviceChannels[1] * audio::getFormatBytes(m_deviceFormat[1]);
-			if (m_mode == audio::orchestra::mode_output && m_deviceBuffer) {
-				uint64_t bytesOut = m_nDeviceChannels[0] * audio::getFormatBytes(m_deviceFormat[0]);
-				if (bufferBytes < bytesOut) {
-					makeBuffer = false;
-				}
-			}
-		}
-		if (makeBuffer) {
-			bufferBytes *= *_bufferSize;
-			if (m_deviceBuffer) free(m_deviceBuffer);
-			m_deviceBuffer = (char *) calloc(bufferBytes, 1);
-			if (m_deviceBuffer == null) {
-				ATA_ERROR("error allocating device buffer memory.");
-				goto error;
-			}
-		}
-	}
-	// 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)] == null)	{
-		ATA_ERROR("error allocating port memory.");
-		goto error;
-	}
-	m_device[modeToIdTable(_mode)] = _device;
-	m_channelOffset[modeToIdTable(_mode)] = _firstChannel;
-	m_state = audio::orchestra::state::stopped;
-	if (    m_mode == audio::orchestra::mode_output
-	     && _mode == audio::orchestra::mode_input) {
-		// We had already set up the stream for output.
-		m_mode = audio::orchestra::mode_duplex;
-	} else {
-		m_mode = _mode;
-		jack_set_process_callback(m_private->client, &audio::orchestra::api::Jack::jackCallbackHandler, this);
-		jack_set_xrun_callback(m_private->client, &audio::orchestra::api::Jack::jackXrun, this);
-		jack_on_shutdown(m_private->client, &audio::orchestra::api::Jack::jackShutdown, this);
-	}
-	// Register our ports.
-	char label[64];
-	if (_mode == audio::orchestra::mode_output) {
-		for (uint32_t i=0; i<m_nUserChannels[0]; i++) {
-			snprintf(label, 64, "outport %d", i);
-			m_private->ports[0][i] = jack_port_register(m_private->client,
-			                                            (const char *)label,
-			                                            JACK_DEFAULT_AUDIO_TYPE,
-			                                            JackPortIsOutput,
-			                                            0);
-		}
-	} else {
-		for (uint32_t i=0; i<m_nUserChannels[1]; i++) {
-			snprintf(label, 64, "inport %d", i);
-			m_private->ports[1][i] = jack_port_register(m_private->client,
-			                                            (const char *)label,
-			                                            JACK_DEFAULT_AUDIO_TYPE,
-			                                            JackPortIsInput,
-			                                            0);
-		}
-	}
-	// Setup the buffer conversion information structure.	We don't use
-	// buffers to do channel offsets, so we override that parameter
-	// here.
-	if (m_doConvertBuffer[modeToIdTable(_mode)]) {
-		setConvertInfo(_mode, 0);
-	}
-	return true;
-error:
-	jack_client_close(m_private->client);
-	if (m_private->ports[0] != null) {
-		free(m_private->ports[0]);
-		m_private->ports[0] = null;
-	}
-	if (m_private->ports[1] != null) {
-		free(m_private->ports[1]);
-		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 = null;
-	}
-	return false;
-}
-
-enum audio::orchestra::error audio::orchestra::api::Jack::closeStream() {
-	if (m_state == audio::orchestra::state::closed) {
-		ATA_ERROR("no open stream to close!");
-		return audio::orchestra::error_warning;
-	}
-	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] != null) {
-		free(m_private->ports[0]);
-		m_private->ports[0] = null;
-	}
-	if (m_private->ports[1] != null) {
-		free(m_private->ports[1]);
-		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 = null;
-	}
-	m_mode = audio::orchestra::mode_unknow;
-	m_state = audio::orchestra::state::closed;
-	return audio::orchestra::error_none;
-}
-
-enum audio::orchestra::error audio::orchestra::api::Jack::startStream() {
-	// TODO : Check return ...
-	audio::orchestra::Api::startStream();
-	if (verifyStream() != audio::orchestra::error_none) {
-		return audio::orchestra::error_fail;
-	}
-	if (m_state == audio::orchestra::state::running) {
-		ATA_ERROR("the stream is already running!");
-		return audio::orchestra::error_warning;
-	}
-	int32_t result = jack_activate(m_private->client);
-	if (result) {
-		ATA_ERROR("unable to activate JACK client!");
-		goto unlock;
-	}
-	const char **ports;
-	// Get the list of available ports.
-	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(), null, JackPortIsInput);
-		if (ports == null) {
-			ATA_ERROR("error determining available JACK input ports!");
-			goto unlock;
-		}
-		// Now make the port connections.	Since RtAudio wasn't designed to
-		// allow the user to select particular channels of a device, we'll
-		// just open the first "nChannels" ports with offset.
-		for (uint32_t i=0; i<m_nUserChannels[0]; i++) {
-			result = 1;
-			if (ports[ m_channelOffset[0] + i ])
-				result = jack_connect(m_private->client, jack_port_name(m_private->ports[0][i]), ports[ m_channelOffset[0] + i ]);
-			if (result) {
-				free(ports);
-				ATA_ERROR("error connecting output ports!");
-				goto unlock;
-			}
-		}
-		free(ports);
-	}
-	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(), null, JackPortIsOutput);
-		if (ports == null) {
-			ATA_ERROR("error determining available JACK output ports!");
-			goto unlock;
-		}
-		// Now make the port connections. See note above.
-		for (uint32_t i=0; i<m_nUserChannels[1]; i++) {
-			result = 1;
-			if (ports[ m_channelOffset[1] + i ]) {
-				result = jack_connect(m_private->client, ports[ m_channelOffset[1] + i ], jack_port_name(m_private->ports[1][i]));
-			}
-			if (result) {
-				free(ports);
-				ATA_ERROR("error connecting input ports!");
-				goto unlock;
-			}
-		}
-		free(ports);
-	}
-	m_private->drainCounter = 0;
-	m_private->internalDrain = false;
-	m_state = audio::orchestra::state::running;
-unlock:
-	if (result == 0) {
-		return audio::orchestra::error_none;
-	}
-	return audio::orchestra::error_systemError;
-}
-
-enum audio::orchestra::error audio::orchestra::api::Jack::stopStream() {
-	if (verifyStream() != audio::orchestra::error_none) {
-		return audio::orchestra::error_fail;
-	}
-	if (m_state == audio::orchestra::state::stopped) {
-		ATA_ERROR("the stream is already stopped!");
-		return audio::orchestra::error_warning;
-	}
-	if (    m_mode == audio::orchestra::mode_output
-	     || m_mode == audio::orchestra::mode_duplex) {
-		if (m_private->drainCounter == 0) {
-			m_private->drainCounter = 2;
-			m_private->m_semaphore.wait();
-		}
-	}
-	jack_deactivate(m_private->client);
-	m_state = audio::orchestra::state::stopped;
-	return audio::orchestra::error_none;
-}
-
-enum audio::orchestra::error audio::orchestra::api::Jack::abortStream() {
-	if (verifyStream() != audio::orchestra::error_none) {
-		return audio::orchestra::error_fail;
-	}
-	if (m_state == audio::orchestra::state::stopped) {
-		ATA_ERROR("the stream is already stopped!");
-		return audio::orchestra::error_warning;
-	}
-	m_private->drainCounter = 2;
-	return stopStream();
-}
-
-bool audio::orchestra::api::Jack::callbackEvent(uint64_t _nframes) {
-	if (    m_state == audio::orchestra::state::stopped
-	     || m_state == audio::orchestra::state::stopping) {
-		return true;
-	}
-	if (m_state == audio::orchestra::state::closed) {
-		ATA_ERROR("the stream is closed ... this shouldn't happen!");
-		return false;
-	}
-	if (m_bufferSize != _nframes) {
-		ATA_ERROR("the JACK buffer size has changed ... cannot process!");
-		return false;
-	}
-	// Check if we were draining the stream and signal is finished.
-	if (m_private->drainCounter > 3) {
-		m_state = audio::orchestra::state::stopping;
-		if (m_private->internalDrain == true) {
-			ETK_NEW(ethread::Thread, [&](){stopStream();}, "Jack_stopStream");
-		} else {
-			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();
-		etk::Vector<enum audio::orchestra::status> status;
-		if (m_mode != audio::orchestra::mode_input && m_private->xrun[0] == true) {
-			status.pushBack(audio::orchestra::status::underflow);
-			m_private->xrun[0] = false;
-		}
-		if (m_mode != audio::orchestra::mode_output && m_private->xrun[1] == true) {
-			status.pushBack(audio::orchestra::status::overflow);
-			m_private->xrun[1] = false;
-		}
-		int32_t cbReturnValue = m_callback(&m_userBuffer[1][0],
-		                                   streamTime,
-		                                   &m_userBuffer[0][0],
-		                                   streamTime,
-		                                   m_bufferSize,
-		                                   status);
-		if (cbReturnValue == 2) {
-			m_state = audio::orchestra::state::stopping;
-			m_private->drainCounter = 2;
-			ETK_NEW(ethread::Thread, [&](){stopStream();}, "Jack_stopStream2");
-			return true;
-		}
-		else if (cbReturnValue == 1) {
-			m_private->drainCounter = 1;
-			m_private->internalDrain = true;
-		}
-	}
-	jack_default_audio_sample_t *jackbuffer;
-	uint64_t bufferBytes = _nframes * sizeof(jack_default_audio_sample_t);
-	if (    m_mode == audio::orchestra::mode_output
-	     || m_mode == audio::orchestra::mode_duplex) {
-		if (m_private->drainCounter > 1) { // write zeros to the output stream
-			for (uint32_t i=0; i<m_nDeviceChannels[0]; i++) {
-				jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer(m_private->ports[0][i], (jack_nframes_t) _nframes);
-				memset(jackbuffer, 0, bufferBytes);
-			}
-		} else if (m_doConvertBuffer[0]) {
-			convertBuffer(m_deviceBuffer, &m_userBuffer[0][0], m_convertInfo[0]);
-			for (uint32_t i=0; i<m_nDeviceChannels[0]; i++) {
-				jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer(m_private->ports[0][i], (jack_nframes_t) _nframes);
-				memcpy(jackbuffer, &m_deviceBuffer[i*bufferBytes], bufferBytes);
-			}
-		} else { // no buffer conversion
-			for (uint32_t i=0; i<m_nUserChannels[0]; i++) {
-				jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer(m_private->ports[0][i], (jack_nframes_t) _nframes);
-				memcpy(jackbuffer, &m_userBuffer[0][i*bufferBytes], bufferBytes);
-			}
-		}
-		if (m_private->drainCounter) {
-			m_private->drainCounter++;
-			goto unlock;
-		}
-	}
-	if (    m_mode == audio::orchestra::mode_input
-	     || m_mode == audio::orchestra::mode_duplex) {
-		if (m_doConvertBuffer[1]) {
-			for (uint32_t i=0; i<m_nDeviceChannels[1]; i++) {
-				jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer(m_private->ports[1][i], (jack_nframes_t) _nframes);
-				memcpy(&m_deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes);
-			}
-			convertBuffer(&m_userBuffer[1][0], m_deviceBuffer, m_convertInfo[1]);
-		} else {
-			// no buffer conversion
-			for (uint32_t i=0; i<m_nUserChannels[1]; i++) {
-				jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer(m_private->ports[1][i], (jack_nframes_t) _nframes);
-				memcpy(&m_userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes);
-			}
-		}
-	}
-unlock:
-	audio::orchestra::Api::tickStreamTime();
-	return true;
-}
-
-#endif
-

audio/orchestra/api/Jack.hpp

@@ -1,56 +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
-#ifdef ORCHESTRA_BUILD_JACK
-
-#include <jack/jack.h>
-
-namespace audio {
-	namespace orchestra {
-		namespace api {
-			class JackPrivate;
-			class Jack: public audio::orchestra::Api {
-				public:
-					static ememory::SharedPtr<audio::orchestra::Api> create();
-				public:
-					Jack();
-					virtual ~Jack();
-					const etk::String& getCurrentApi() {
-						return audio::orchestra::typeJack;
-					}
-					uint32_t getDeviceCount();
-					audio::orchestra::DeviceInfo getDeviceInfo(uint32_t _device);
-					enum audio::orchestra::error closeStream();
-					enum audio::orchestra::error startStream();
-					enum audio::orchestra::error stopStream();
-					enum audio::orchestra::error abortStream();
-					long getStreamLatency();
-					// This function is intended for internal use only.	It must be
-					// public because it is called by the internal callback handler,
-					// which is not a member of RtAudio.	External use of this function
-					// will most likely produce highly undesireable results!
-					bool callbackEvent(uint64_t _nframes);
-				private:
-					static int32_t jackXrun(void* _userData);
-					static void jackShutdown(void* _userData);
-					static int32_t jackCallbackHandler(jack_nframes_t _nframes, void* _userData);
-				private:
-					ememory::SharedPtr<JackPrivate> m_private;
-					bool open(uint32_t _device,
-					          audio::orchestra::mode _mode,
-					          uint32_t _channels,
-					          uint32_t _firstChannel,
-					          uint32_t _sampleRate,
-					          audio::format _format,
-					          uint32_t *_bufferSize,
-					          const audio::orchestra::StreamOptions& _options);
-			};
-		}
-	}
-}
-
-#endif

audio/orchestra/api/Pulse.cpp

@@ -1,412 +0,0 @@
-/** @file
- * @author Edouard DUPIN 
- * @copyright 2011, Edouard DUPIN, all right reserved
- * @license APACHE v2.0 (see license file)
- * @fork from RTAudio
- */
-
-
-#if defined(ORCHESTRA_BUILD_PULSE)
-
-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 <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>(ETK_NEW(audio::orchestra::api::Pulse));
-}
-
-
-static const uint32_t SUPPORTED_SAMPLERATES[] = {
-	8000,
-	16000,
-	22050,
-	32000,
-	44100,
-	48000,
-	96000,
-	0
-};
-
-struct rtaudio_pa_format_mapping_t {
-	enum audio::format airtaudio_format;
-	pa_sample_format_t pa_format;
-};
-
-static const rtaudio_pa_format_mapping_t supported_sampleformats[] = {
-	{audio::format_int16, PA_SAMPLE_S16LE},
-	{audio::format_int32, PA_SAMPLE_S32LE},
-	{audio::format_float, PA_SAMPLE_FLOAT32LE},
-	{audio::format_unknow, PA_SAMPLE_INVALID}};
-
-
-namespace audio {
-	namespace orchestra {
-		namespace api {
-			class PulsePrivate {
-				public:
-					pa_simple* handle;
-					ememory::SharedPtr<ethread::Thread> thread;
-					bool threadRunning;
-					ethread::Semaphore m_semaphore;
-					bool runnable;
-					PulsePrivate() :
-					  handle(0),
-					  threadRunning(false),
-					  runnable(false) {
-						
-					}
-			};
-		}
-	}
-}
-audio::orchestra::api::Pulse::Pulse() :
-  m_private(ETK_NEW(audio::orchestra::api::PulsePrivate)) {
-	
-}
-
-audio::orchestra::api::Pulse::~Pulse() {
-	if (m_state != audio::orchestra::state::closed) {
-		closeStream();
-	}
-}
-
-uint32_t audio::orchestra::api::Pulse::getDeviceCount() {
-	#if 1
-		etk::Vector<audio::orchestra::DeviceInfo> list = audio::orchestra::api::pulse::getDeviceList();
-		return list.size();
-	#else
-		return 1;
-	#endif
-}
-
-audio::orchestra::DeviceInfo audio::orchestra::api::Pulse::getDeviceInfo(uint32_t _device) {
-	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();
-	}
-	return list[_device];
-}
-
-
-
-void audio::orchestra::api::Pulse::callbackEvent() {
-	ethread::setName("Pulse IO-" + m_name);
-	while (m_private->threadRunning == true) {
-		callbackEventOneCycle();
-	}
-}
-
-enum audio::orchestra::error audio::orchestra::api::Pulse::closeStream() {
-	m_private->threadRunning = false;
-	m_mutex.lock();
-	if (m_state == audio::orchestra::state::stopped) {
-		m_private->runnable = true;
-		m_private->m_semaphore.post();;
-	}
-	m_mutex.unLock();
-	m_private->thread->join();
-	if (m_mode == audio::orchestra::mode_output) {
-		pa_simple_flush(m_private->handle, null);
-	}
-	pa_simple_free(m_private->handle);
-	m_private->handle = null;
-	m_userBuffer[0].clear();
-	m_userBuffer[1].clear();
-	m_state = audio::orchestra::state::closed;
-	m_mode = audio::orchestra::mode_unknow;
-	return audio::orchestra::error_none;
-}
-
-void audio::orchestra::api::Pulse::callbackEventOneCycle() {
-	if (m_state == audio::orchestra::state::stopped) {
-		while (!m_private->runnable) {
-			m_private->m_semaphore.wait();
-		}
-		if (m_state != audio::orchestra::state::running) {
-			m_mutex.unLock();
-			return;
-		}
-	}
-	if (m_state == audio::orchestra::state::closed) {
-		ATA_ERROR("the stream is closed ... this shouldn't happen!");
-		return;
-	}
-	audio::Time streamTime = getStreamTime();
-	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],
-	                                  streamTime,
-	                                  m_bufferSize,
-	                                  status);
-	if (doStopStream == 2) {
-		abortStream();
-		return;
-	}
-	m_mutex.lock();
-	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;
-	}
-	int32_t pa_error;
-	size_t bytes;
-	if (m_mode == audio::orchestra::mode_output) {
-		if (m_doConvertBuffer[audio::orchestra::modeToIdTable(audio::orchestra::mode_output)]) {
-			convertBuffer(m_deviceBuffer,
-			              &m_userBuffer[audio::orchestra::modeToIdTable(audio::orchestra::mode_output)][0],
-			              m_convertInfo[audio::orchestra::modeToIdTable(audio::orchestra::mode_output)]);
-			bytes = m_nDeviceChannels[audio::orchestra::modeToIdTable(audio::orchestra::mode_output)] * m_bufferSize * audio::getFormatBytes(m_deviceFormat[audio::orchestra::modeToIdTable(audio::orchestra::mode_output)]);
-		} else {
-			bytes = m_nUserChannels[audio::orchestra::modeToIdTable(audio::orchestra::mode_output)] * m_bufferSize * audio::getFormatBytes(m_userFormat);
-		}
-		if (pa_simple_write(m_private->handle, pulse_out, bytes, &pa_error) < 0) {
-			ATA_ERROR("audio write error, " << pa_strerror(pa_error) << ".");
-			return;
-		}
-	}
-	if (m_mode == audio::orchestra::mode_input) {
-		if (m_doConvertBuffer[audio::orchestra::modeToIdTable(audio::orchestra::mode_input)]) {
-			bytes = m_nDeviceChannels[audio::orchestra::modeToIdTable(audio::orchestra::mode_input)] * m_bufferSize * audio::getFormatBytes(m_deviceFormat[audio::orchestra::modeToIdTable(audio::orchestra::mode_input)]);
-		} else {
-			bytes = m_nUserChannels[audio::orchestra::modeToIdTable(audio::orchestra::mode_input)] * m_bufferSize * audio::getFormatBytes(m_userFormat);
-		}
-		if (pa_simple_read(m_private->handle, pulse_in, bytes, &pa_error) < 0) {
-			ATA_ERROR("audio read error, " << pa_strerror(pa_error) << ".");
-			return;
-		}
-		if (m_doConvertBuffer[audio::orchestra::modeToIdTable(audio::orchestra::mode_input)]) {
-			convertBuffer(&m_userBuffer[audio::orchestra::modeToIdTable(audio::orchestra::mode_input)][0],
-			              m_deviceBuffer,
-			              m_convertInfo[audio::orchestra::modeToIdTable(audio::orchestra::mode_input)]);
-		}
-	}
-unLock:
-	m_mutex.unLock();
-	audio::orchestra::Api::tickStreamTime();
-	if (doStopStream == 1) {
-		stopStream();
-		return;
-	}
-	return;
-}
-
-enum audio::orchestra::error audio::orchestra::api::Pulse::startStream() {
-	// TODO : Check return ...
-	audio::orchestra::Api::startStream();
-	if (m_state == audio::orchestra::state::closed) {
-		ATA_ERROR("the stream is not open!");
-		return audio::orchestra::error_invalidUse;
-	}
-	if (m_state == audio::orchestra::state::running) {
-		ATA_ERROR("the stream is already running!");
-		return audio::orchestra::error_warning;
-	}
-	m_mutex.lock();
-	m_state = audio::orchestra::state::running;
-	m_private->runnable = true;
-	m_private->m_semaphore.post();
-	m_mutex.unLock();
-	return audio::orchestra::error_none;
-}
-
-enum audio::orchestra::error audio::orchestra::api::Pulse::stopStream() {
-	if (m_state == audio::orchestra::state::closed) {
-		ATA_ERROR("the stream is not open!");
-		return audio::orchestra::error_invalidUse;
-	}
-	if (m_state == audio::orchestra::state::stopped) {
-		ATA_ERROR("the stream is already stopped!");
-		return audio::orchestra::error_warning;
-	}
-	m_state = audio::orchestra::state::stopped;
-	m_mutex.lock();
-	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();
-			return audio::orchestra::error_systemError;
-		}
-	}
-	m_state = audio::orchestra::state::stopped;
-	m_mutex.unLock();
-	return audio::orchestra::error_none;
-}
-
-enum audio::orchestra::error audio::orchestra::api::Pulse::abortStream() {
-	if (m_state == audio::orchestra::state::closed) {
-		ATA_ERROR("the stream is not open!");
-		return audio::orchestra::error_invalidUse;
-	}
-	if (m_state == audio::orchestra::state::stopped) {
-		ATA_ERROR("the stream is already stopped!");
-		return audio::orchestra::error_warning;
-	}
-	m_state = audio::orchestra::state::stopped;
-	m_mutex.lock();
-	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();
-			return audio::orchestra::error_systemError;
-		}
-	}
-	m_state = audio::orchestra::state::stopped;
-	m_mutex.unLock();
-	return audio::orchestra::error_none;
-}
-
-bool audio::orchestra::api::Pulse::open(uint32_t _device,
-                                        audio::orchestra::mode _mode,
-                                        uint32_t _channels,
-                                        uint32_t _firstChannel,
-                                        uint32_t _sampleRate,
-                                        audio::format _format,
-                                        uint32_t *_bufferSize,
-                                        const audio::orchestra::StreamOptions& _options) {
-	uint64_t bufferBytes = 0;
-	pa_sample_spec ss;
-	if (_device != 0) {
-		return false;
-	}
-	if (_mode != audio::orchestra::mode_input && _mode != audio::orchestra::mode_output) {
-		return false;
-	}
-	if (_channels != 1 && _channels != 2) {
-		ATA_ERROR("unsupported number of channels.");
-		return false;
-	}
-	ss.channels = _channels;
-	if (_firstChannel != 0) {
-		return false;
-	}
-	bool sr_found = false;
-	for (const uint32_t *sr = SUPPORTED_SAMPLERATES; *sr; ++sr) {
-		if (_sampleRate == *sr) {
-			sr_found = true;
-			m_sampleRate = _sampleRate;
-			ss.rate = _sampleRate;
-			break;
-		}
-	}
-	if (!sr_found) {
-		ATA_ERROR("unsupported sample rate.");
-		return false;
-	}
-	bool sf_found = 0;
-	for (const rtaudio_pa_format_mapping_t *sf = supported_sampleformats;
-	     sf->airtaudio_format && sf->pa_format != PA_SAMPLE_INVALID;
-	     ++sf) {
-		if (_format == sf->airtaudio_format) {
-			sf_found = true;
-			m_userFormat = sf->airtaudio_format;
-			ss.format = sf->pa_format;
-			break;
-		}
-	}
-	if (!sf_found) {
-		ATA_ERROR("unsupported sample format.");
-		return false;
-	}
-	m_deviceInterleaved[modeToIdTable(_mode)] = true;
-	m_nBuffers = 1;
-	m_doByteSwap[modeToIdTable(_mode)] = false;
-	m_doConvertBuffer[modeToIdTable(_mode)] = false;
-	m_deviceFormat[modeToIdTable(_mode)] = m_userFormat;
-	m_nUserChannels[modeToIdTable(_mode)] = _channels;
-	m_nDeviceChannels[modeToIdTable(_mode)] = _channels + _firstChannel;
-	m_channelOffset[modeToIdTable(_mode)] = 0;
-	// Allocate necessary internal buffers.
-	bufferBytes = m_nUserChannels[modeToIdTable(_mode)] * *_bufferSize * audio::getFormatBytes(m_userFormat);
-	m_userBuffer[modeToIdTable(_mode)].resize(bufferBytes, 0);
-	if (m_userBuffer[modeToIdTable(_mode)].size() == 0) {
-		ATA_ERROR("error allocating user buffer memory.");
-		goto error;
-	}
-	m_bufferSize = *_bufferSize;
-	if (m_doConvertBuffer[modeToIdTable(_mode)]) {
-		bool makeBuffer = true;
-		bufferBytes = m_nDeviceChannels[modeToIdTable(_mode)] * audio::getFormatBytes(m_deviceFormat[modeToIdTable(_mode)]);
-		if (_mode == audio::orchestra::mode_input) {
-			if (m_mode == audio::orchestra::mode_output && m_deviceBuffer) {
-				uint64_t bytesOut = m_nDeviceChannels[0] * audio::getFormatBytes(m_deviceFormat[0]);
-				if (bufferBytes <= bytesOut) makeBuffer = false;
-			}
-		}
-		if (makeBuffer) {
-			bufferBytes *= *_bufferSize;
-			if (m_deviceBuffer) free(m_deviceBuffer);
-			m_deviceBuffer = (char *) calloc(bufferBytes, 1);
-			if (m_deviceBuffer == null) {
-				ATA_ERROR("error allocating device buffer memory.");
-				goto error;
-			}
-		}
-	}
-	m_device[modeToIdTable(_mode)] = _device;
-	// Setup the buffer conversion information structure.
-	if (m_doConvertBuffer[modeToIdTable(_mode)]) {
-		setConvertInfo(_mode, _firstChannel);
-	}
-	int32_t error;
-	switch (_mode) {
-		case audio::orchestra::mode_input:
-			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(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;
-			}
-			break;
-		default:
-			goto error;
-	}
-	if (m_mode == audio::orchestra::mode_unknow) {
-		m_mode = _mode;
-	} else {
-		goto error;
-	}
-	if (m_private->threadRunning == false) {
-		m_private->threadRunning = true;
-		m_private->thread = ememory::makeShared<ethread::Thread>([&](){callbackEvent();}, "pulseCallback");
-		if (m_private->thread == null) {
-			ATA_ERROR("error creating thread.");
-			goto error;
-		}
-	}
-	m_state = audio::orchestra::state::stopped;
-	return true;
-error:
-	for (int32_t iii=0; iii<2; ++iii) {
-		m_userBuffer[iii].clear();
-	}
-	if (m_deviceBuffer) {
-		free(m_deviceBuffer);
-		m_deviceBuffer = 0;
-	}
-	return false;
-}
-
-#endif

audio/orchestra/api/Pulse.hpp

@@ -1,52 +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
-#ifdef ORCHESTRA_BUILD_PULSE
-
-namespace audio {
-	namespace orchestra {
-		namespace api {
-			class PulsePrivate;
-			class Pulse: public audio::orchestra::Api {
-				public:
-					static ememory::SharedPtr<audio::orchestra::Api> create();
-				public:
-					Pulse();
-					virtual ~Pulse();
-					const etk::String& getCurrentApi() {
-						return audio::orchestra::typePulse;
-					}
-					uint32_t getDeviceCount();
-					audio::orchestra::DeviceInfo getDeviceInfo(uint32_t _device);
-					enum audio::orchestra::error closeStream();
-					enum audio::orchestra::error startStream();
-					enum audio::orchestra::error stopStream();
-					enum audio::orchestra::error abortStream();
-					// This function is intended for internal use only.	It must be
-					// public because it is called by the internal callback handler,
-					// which is not a member of RtAudio.	External use of this function
-					// will most likely produce highly undesireable results!
-					void callbackEventOneCycle();
-					void callbackEvent();
-				private:
-					ememory::SharedPtr<PulsePrivate> m_private;
-					etk::Vector<audio::orchestra::DeviceInfo> m_devices;
-					void saveDeviceInfo();
-					bool open(uint32_t _device,
-					          audio::orchestra::mode _mode,
-					          uint32_t _channels,
-					          uint32_t _firstChannel,
-					          uint32_t _sampleRate,
-					          audio::format _format,
-					          uint32_t *_bufferSize,
-					          const audio::orchestra::StreamOptions& _options);
-			};
-		}
-	}
-}
-
-#endif

audio/orchestra/api/PulseDeviceList.cpp

@@ -1,363 +0,0 @@
-/** @file
- * @author Edouard DUPIN 
- * @copyright 2011, Edouard DUPIN, all right reserved
- * @license APACHE v2.0 (see license file)
- * @fork from RTAudio
- */
-
-#if defined(ORCHESTRA_BUILD_PULSE)
-extern "C" {
-	#include <stdio.h>
-	#include <string.h>
-}
-#include <pulse/pulseaudio.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
-static void callbackStateMachine(pa_context* _contex, void *_userdata) {
-	pa_context_state_t state;
-	int *pulseAudioReady = static_cast<int*>(_userdata);
-	state = pa_context_get_state(_contex);
-	switch  (state) {
-		// There are just here for reference
-		case PA_CONTEXT_UNCONNECTED:
-			ATA_VERBOSE("pulse state: PA_CONTEXT_UNCONNECTED");
-			break;
-		case PA_CONTEXT_CONNECTING:
-			ATA_VERBOSE("pulse state: PA_CONTEXT_CONNECTING");
-			break;
-		case PA_CONTEXT_AUTHORIZING:
-			ATA_VERBOSE("pulse state: PA_CONTEXT_AUTHORIZING");
-			break;
-		case PA_CONTEXT_SETTING_NAME:
-			ATA_VERBOSE("pulse state: PA_CONTEXT_SETTING_NAME");
-			break;
-		default:
-			ATA_VERBOSE("pulse state: default");
-			break;
-		case PA_CONTEXT_FAILED:
-			*pulseAudioReady = 2;
-			ATA_VERBOSE("pulse state: PA_CONTEXT_FAILED");
-			break;
-		case PA_CONTEXT_TERMINATED:
-			*pulseAudioReady = 2;
-			ATA_VERBOSE("pulse state: PA_CONTEXT_TERMINATED");
-			break;
-		case PA_CONTEXT_READY:
-			*pulseAudioReady = 1;
-			ATA_VERBOSE("pulse state: PA_CONTEXT_READY");
-			break;
-	}
-}
-
-static audio::format getFormatFromPulseFormat(enum pa_sample_format _format) {
-	switch (_format) {
-		case PA_SAMPLE_U8:
-			return audio::format_int8;
-			break;
-		case PA_SAMPLE_ALAW:
-			ATA_ERROR("Not supported: uint8_t a-law");
-			return audio::format_unknow;
-		case PA_SAMPLE_ULAW:
-			ATA_ERROR("Not supported: uint8_t mu-law");
-			return audio::format_unknow;
-		case PA_SAMPLE_S16LE:
-			return audio::format_int16;
-			break;
-		case PA_SAMPLE_S16BE:
-			return audio::format_int16;
-			break;
-		case PA_SAMPLE_FLOAT32LE:
-			return audio::format_float;
-			break;
-		case PA_SAMPLE_FLOAT32BE:
-			return audio::format_float;
-			break;
-		case PA_SAMPLE_S32LE:
-			return audio::format_int32;
-			break;
-		case PA_SAMPLE_S32BE:
-			return audio::format_int32;
-			break;
-		case PA_SAMPLE_S24LE:
-			return audio::format_int24;
-			break;
-		case PA_SAMPLE_S24BE:
-			return audio::format_int24;
-			break;
-		case PA_SAMPLE_S24_32LE:
-			return audio::format_int24_on_int32;
-			break;
-		case PA_SAMPLE_S24_32BE:
-			return audio::format_int24_on_int32;
-			break;
-		case PA_SAMPLE_INVALID:
-		case PA_SAMPLE_MAX:
-			ATA_ERROR("Not supported: invalid");
-			return audio::format_unknow;
-	}
-	ATA_ERROR("Not supported: UNKNOW flag...");
-	return audio::format_unknow;
-}
-
-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.pushBack(audio::channel_unknow);
-				break;
-			case PA_CHANNEL_POSITION_MONO:
-			case PA_CHANNEL_POSITION_FRONT_CENTER:
-				out.pushBack(audio::channel_frontCenter);
-				break;
-			case PA_CHANNEL_POSITION_FRONT_LEFT:
-				out.pushBack(audio::channel_frontLeft);
-				break;
-			case PA_CHANNEL_POSITION_FRONT_RIGHT:
-				out.pushBack(audio::channel_frontRight);
-				break;
-			case PA_CHANNEL_POSITION_REAR_CENTER:
-				out.pushBack(audio::channel_rearCenter);
-				break;
-			case PA_CHANNEL_POSITION_REAR_LEFT:
-				out.pushBack(audio::channel_rearLeft);
-				break;
-			case PA_CHANNEL_POSITION_REAR_RIGHT:
-				out.pushBack(audio::channel_rearRight);
-				break;
-			case PA_CHANNEL_POSITION_LFE:
-				out.pushBack(audio::channel_lfe);
-				break;
-			case PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER:
-				out.pushBack(audio::channel_centerLeft);
-				break;
-			case PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER:
-				out.pushBack(audio::channel_centerRight);
-				break;
-			case PA_CHANNEL_POSITION_SIDE_LEFT:
-				out.pushBack(audio::channel_topCenterLeft);
-				break;
-			case PA_CHANNEL_POSITION_SIDE_RIGHT:
-				out.pushBack(audio::channel_topCenterRight);
-				break;
-			case PA_CHANNEL_POSITION_TOP_CENTER:
-			case PA_CHANNEL_POSITION_TOP_FRONT_CENTER:
-				out.pushBack(audio::channel_topFrontCenter);
-				break;
-			case PA_CHANNEL_POSITION_TOP_FRONT_LEFT:
-				out.pushBack(audio::channel_topFrontLeft);
-				break;
-			case PA_CHANNEL_POSITION_TOP_FRONT_RIGHT:
-				out.pushBack(audio::channel_topFrontRight);
-				break;
-			case PA_CHANNEL_POSITION_TOP_REAR_LEFT:
-				out.pushBack(audio::channel_topRearLeft);
-				break;
-			case PA_CHANNEL_POSITION_TOP_REAR_RIGHT:
-				out.pushBack(audio::channel_topRearRight);
-				break;
-			case PA_CHANNEL_POSITION_TOP_REAR_CENTER:
-				out.pushBack(audio::channel_topRearCenter);
-				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;
-		}
-	}
-	
-	return out;
-}
-// Callback on getting data from pulseaudio:
-static void callbackGetSinkList(pa_context* _contex, const pa_sink_info* _info, int _eol, void* _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;
-	}
-	audio::orchestra::DeviceInfo info;
-	info.isCorrect = true;
-	info.input = false;
-	info.name = _info->name;
-	info.desc = _info->description;
-	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->pushBack(info);
-}
-
-// allback to get data from pulseaudio:
-static void callbackGetSourceList(pa_context* _contex, const pa_source_info* _info, int _eol, void* _userdata) {
-	etk::Vector<audio::orchestra::DeviceInfo>* list = static_cast<etk::Vector<audio::orchestra::DeviceInfo>*>(_userdata);
-	if (_eol > 0) {
-		return;
-	}
-	audio::orchestra::DeviceInfo info;
-	info.isCorrect = true;
-	info.input = true;
-	info.name = _info->name;
-	info.desc = _info->description;
-	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->pushBack(info);
-}
-
-// to not update all the time ...
-static etk::Vector<audio::orchestra::DeviceInfo> pulseAudioListOfDevice;
-static audio::Time pulseAudioListOfDeviceTime;
-
-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;
-	}
-	// Define our pulse audio loop and connection variables
-	pa_mainloop* pulseAudioMainLoop;
-	pa_mainloop_api* pulseAudioMainLoopAPI;
-	pa_operation* pulseAudioOperation;
-	pa_context* pulseAudioContex;
-	pa_context_flags_t pulseAudioFlags = PA_CONTEXT_NOAUTOSPAWN;
-	etk::Vector<audio::orchestra::DeviceInfo>& out = pulseAudioListOfDevice;
-	out.clear();
-	// We'll need these state variables to keep track of our requests
-	int state = 0;
-	int pulseAudioReady = 0;
-	// Create a mainloop API and connection to the default server
-	pulseAudioMainLoop = pa_mainloop_new();
-	pulseAudioMainLoopAPI = pa_mainloop_get_api(pulseAudioMainLoop);
-	pulseAudioContex = pa_context_new(pulseAudioMainLoopAPI, "orchestraPulseCount");
-	// If there's an error, the callback will set pulseAudioReady
-	pa_context_set_state_callback(pulseAudioContex, callbackStateMachine, &pulseAudioReady);
-	// This function connects to the pulse server
-	pa_context_connect(pulseAudioContex, NULL, pulseAudioFlags, NULL);
-	bool playLoop = true;
-	while (playLoop == true) {
-		// We can't do anything until PA is ready, so just iterate the mainloop
-		// and continue
-		if (pulseAudioReady == 0) {
-			pa_mainloop_iterate(pulseAudioMainLoop, 1, null);
-			continue;
-		}
-		// We couldn't get a connection to the server, so exit out
-		if (pulseAudioReady == 2) {
-			pa_context_disconnect(pulseAudioContex);
-			pa_context_unref(pulseAudioContex);
-			pa_mainloop_free(pulseAudioMainLoop);
-			ATA_ERROR("Pulse interface error: Can not connect to the pulseaudio iterface...");
-			return out;
-		}
-		// At this point, we're connected to the server and ready to make
-		// requests
-		switch (state) {
-			// State 0: we haven't done anything yet
-			case 0:
-				ATA_DEBUG("Request sink list");
-				pulseAudioOperation = pa_context_get_sink_info_list(pulseAudioContex,
-				                                                    callbackGetSinkList,
-				                                                    &out);
-				state++;
-				break;
-			case 1:
-				// Now we wait for our operation to complete.  When it's
-				// complete our pa_output_devicelist is filled out, and we move
-				// along to the next state
-				if (pa_operation_get_state(pulseAudioOperation) == PA_OPERATION_DONE) {
-					pa_operation_unref(pulseAudioOperation);
-					ATA_DEBUG("Request sources list");
-					pulseAudioOperation = pa_context_get_source_info_list(pulseAudioContex,
-					                                                      callbackGetSourceList,
-					                                                      &out);
-					state++;
-				}
-				break;
-			case 2:
-				if (pa_operation_get_state(pulseAudioOperation) == PA_OPERATION_DONE) {
-					ATA_DEBUG("All is done");
-					// Now we're done, clean up and disconnect and return
-					pa_operation_unref(pulseAudioOperation);
-					pa_context_disconnect(pulseAudioContex);
-					pa_context_unref(pulseAudioContex);
-					pa_mainloop_free(pulseAudioMainLoop);
-					playLoop = false;
-					break;
-				}
-			break;
-			default:
-				// We should never see this state
-				ATA_ERROR("Error in getting the devices list ...");
-				return out;
-		}
-		// Iterate the main loop ..
-		if (playLoop == true) {
-			pa_mainloop_iterate(pulseAudioMainLoop, 1, null);
-		}
-	}
-	// TODO: need to do it better ...
-	// set default device:
-	int32_t idInput = -1;
-	int32_t idOutput = -1;
-	for (int32_t iii=0; iii<out.size(); ++iii) {
-		if (out[iii].input == true) {
-			if (idInput != -1) {
-				continue;
-			}
-			if (etk::end_with(out[iii].name, ".monitor", false) == false) {
-				idInput = iii;
-				out[iii].isDefault = true;
-			}
-		} else {
-			if (idOutput != -1) {
-				continue;
-			}
-			if (etk::end_with(out[iii].name, ".monitor", false) == false) {
-				idOutput = iii;
-				out[iii].isDefault = true;
-			}
-		}
-	}
-	return out;
-}
-
-#endif

audio/orchestra/api/PulseDeviceList.hpp

@@ -1,23 +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
-#ifdef ORCHESTRA_BUILD_PULSE
-
-#include <etk/types.hpp>
-#include <audio/orchestra/DeviceInfo.hpp>
-
-namespace audio {
-	namespace orchestra {
-		namespace api {
-			namespace pulse {
-				etk::Vector<audio::orchestra::DeviceInfo> getDeviceList();
-			}
-		}
-	}
-}
-
-#endif

audio/orchestra/base.cpp

@@ -1,6 +0,0 @@
-/** @file
- * @author Edouard DUPIN 
- * @copyright 2011, Edouard DUPIN, all right reserved
- * @license APACHE v2.0 (see license file)
- * @fork from RTAudio
- */

audio/orchestra/base.hpp

@@ -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 <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

@@ -1,13 +0,0 @@
-/** @file
- * @author Edouard DUPIN 
- * @copyright 2011, Edouard DUPIN, all right reserved
- * @license APACHE v2.0 (see license file)
- * @fork from RTAudio
- */
-
-#include <audio/orchestra/debug.hpp>
-
-int32_t audio::orchestra::getLogId() {
-	static int32_t g_val = elog::registerInstance("audio-orchestra");
-	return g_val;
-}

audio/orchestra/debug.hpp

@@ -1,41 +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 <elog/log.hpp>
-
-namespace audio {
-	namespace orchestra {
-		int32_t getLogId();
-	}
-}
-#define ATA_BASE(info,data) ELOG_BASE(audio::orchestra::getLogId(),info,data)
-
-#define ATA_PRINT(data)         ATA_BASE(-1, data)
-#define ATA_CRITICAL(data)      ATA_BASE(1, data)
-#define ATA_ERROR(data)         ATA_BASE(2, data)
-#define ATA_WARNING(data)       ATA_BASE(3, data)
-#ifdef DEBUG
-	#define ATA_INFO(data)          ATA_BASE(4, data)
-	#define ATA_DEBUG(data)         ATA_BASE(5, data)
-	#define ATA_VERBOSE(data)       ATA_BASE(6, data)
-	#define ATA_TODO(data)          ATA_BASE(4, "TODO : " << data)
-#else
-	#define ATA_INFO(data)          do { } while(false)
-	#define ATA_DEBUG(data)         do { } while(false)
-	#define ATA_VERBOSE(data)       do { } while(false)
-	#define ATA_TODO(data)          do { } while(false)
-#endif
-
-#define ATA_ASSERT(cond,data) \
-	do { \
-		if (!(cond)) { \
-			ATA_CRITICAL(data); \
-			assert(!#cond); \
-		} \
-	} while (0)
-

audio/orchestra/error.cpp

@@ -1,9 +0,0 @@
-/** @file
- * @author Edouard DUPIN 
- * @copyright 2011, Edouard DUPIN, all right reserved
- * @license APACHE v2.0 (see license file)
- * @fork from RTAudio
- */
-
-#include <audio/orchestra/error.hpp>
-#include <audio/orchestra/debug.hpp>

audio/orchestra/error.hpp

@@ -1,22 +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.hpp>
-
-namespace audio {
-	namespace orchestra {
-		enum error {
-			error_none, //!< No error
-			error_fail, //!< An error occure in the operation
-			error_warning, //!< A non-critical error.
-			error_inputNull, //!< null input or internal errror
-			error_invalidUse, //!< The function was called incorrectly.
-			error_systemError //!< A system error occured.
-		};
-	}
-}

audio/orchestra/mode.cpp

@@ -1,39 +0,0 @@
-/** @file
- * @author Edouard DUPIN 
- * @copyright 2011, Edouard DUPIN, all right reserved
- * @license APACHE v2.0 (see license file)
- * @fork from RTAudio
- */
-
-#include <audio/orchestra/mode.hpp>
-#include <audio/orchestra/debug.hpp>
-
-int32_t audio::orchestra::modeToIdTable(enum mode _mode) {
-	switch (_mode) {
-		case mode_unknow:
-		case mode_duplex:
-		case mode_output:
-			return 0;
-		case mode_input:
-			return 1;
-	}
-	return 0;
-}
-
-etk::Stream& audio::operator <<(etk::Stream& _os, enum audio::orchestra::mode _obj) {
-	switch (_obj) {
-		case audio::orchestra::mode_unknow:
-			_os << "unknow";
-			break;
-		case audio::orchestra::mode_duplex:
-			_os << "duplex";
-			break;
-		case audio::orchestra::mode_output:
-			_os << "output";
-			break;
-		case audio::orchestra::mode_input:
-			_os << "input";
-			break;
-	}
-	return _os;
-}

audio/orchestra/mode.hpp

@@ -1,24 +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.hpp>
-#include <etk/Stream.hpp>
-
-namespace audio {
-	namespace orchestra {
-		enum mode {
-			mode_unknow,
-			mode_output,
-			mode_input,
-			mode_duplex
-		};
-		int32_t modeToIdTable(enum mode _mode);
-	}
-	etk::Stream& operator <<(etk::Stream& _os, enum audio::orchestra::mode _obj);
-}
-

audio/orchestra/state.cpp

@@ -1,6 +0,0 @@
-/** @file
- * @author Edouard DUPIN 
- * @copyright 2011, Edouard DUPIN, all right reserved
- * @license APACHE v2.0 (see license file)
- * @fork from RTAudio
- */

audio/orchestra/state.hpp

@@ -1,21 +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.hpp>
-
-namespace audio {
-	namespace orchestra {
-		enum class state {
-			closed,
-			stopped,
-			stopping,
-			running
-		};
-	}
-}
-

audio/orchestra/status.cpp

@@ -1,32 +0,0 @@
-/** @file
- * @author Edouard DUPIN 
- * @copyright 2011, Edouard DUPIN, all right reserved
- * @license APACHE v2.0 (see license file)
- * @fork from RTAudio
- */
-
-#include <audio/orchestra/status.hpp>
-#include <audio/orchestra/debug.hpp>
-static const char* listValue[] = {
-	"ok",
-	"overflow",
-	"underflow"
-};
-
-etk::Stream& audio::orchestra::operator <<(etk::Stream& _os, enum audio::orchestra::status _obj) {
-	_os << listValue[int32_t(_obj)];
-	return _os;
-}
-
-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 << etk::String(";");
-		}
-		_os << _obj[iii];
-	}
-	_os << etk::String("}");
-	return _os;
-}
-

audio/orchestra/status.hpp

@@ -1,23 +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.hpp>
-#include <etk/Vector.hpp>
-
-namespace audio {
-	namespace orchestra {
-		enum class status {
-			ok, //!< nothing...
-			overflow, //!< Internal buffer has more data than they can accept
-			underflow //!< The internal buffer is empty
-		};
-		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

@@ -1,21 +0,0 @@
-/** @file
- * @author Edouard DUPIN 
- * @copyright 2011, Edouard DUPIN, all right reserved
- * @license APACHE v2.0 (see license file)
- * @fork from RTAudio
- */
-
-#include <audio/orchestra/type.hpp>
-#include <audio/orchestra/debug.hpp>
-
-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.hpp

@@ -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.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).
-	}
-}
-

audio_orchestra_build.html

@@ -0,0 +1,184 @@
+<!-- HTML header for doxygen 1.8.8-->
+<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
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+        <title>Orchestra: audio interface wrapper: Build lib &amp; build sample</title>
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+                        <li class="dropdown">
+                            <a href="#" class="dropdown-toggle" data-toggle="dropdown" role="button" aria-haspopup="true" aria-expanded="false">
+                                Link-libs<span class="caret"></span>
+                            </a>
+                            <ul class="dropdown-menu">
+                                <li><a href="http://HeeroYui.github.io/lutin">lutin</a></li>
+                                <li><a href="http://atria-soft.github.io/ewol">ewol</a></li>
+                                <li><a href="http://atria-soft.github.io/echrono">echrono</a></li>
+                                <li><a href="http://atria-soft.github.io/etk">etk</a></li>
+                                <li><a href="http://atria-soft.github.io/ejson">ejson</a></li>
+                                <li><a href="http://atria-soft.github.io/exml">exml</a></li>
+                                <li><a href="http://atria-soft.github.io/esvg">esvg</a></li>
+                                <li><a href="http://atria-soft.github.io/egami">egami</a></li>
+                                <li><a href="http://atria-soft.github.io/gale">gale</a></li>
+                                <li><a href="http://atria-soft.github.io/ege">ege</a></li>
+                                <li><a href="http://atria-soft.github.io/elog">elog</a></li>
+                                <li><a href="http://atria-soft.github.io/ememory">ememory</a></li>
+                                <li><a href="http://atria-soft.github.io/enet">enet</a></li>
+                                <li><a href="http://atria-soft.github.io/eproperty">eproperty</a></li>
+                                <li><a href="http://atria-soft.github.io/esignal">esignal</a></li>
+                                <li><a href="http://atria-soft.github.io/etranslate">etranslate</a></li>
+                                <li><a href="http://atria-soft.github.io/zeus">zeus</a></li>
+                                <li><a href="http://musicdsp.github.io/audio-ess">audio-ess</a></li>
+                                <li><a href="http://musicdsp.github.io/audio">audio</a></li>
+                                <li><a href="http://musicdsp.github.io/audio-drain">audio-drain</a></li>
+                                <li><a href="http://musicdsp.github.io/audio-orchestra">audio-orchestra</a></li>
+                                <li><a href="http://musicdsp.github.io/audio-river">audio-river</a></li>
+                            </ul>
+                        </li>
+                    </ul>
+                    <div id="search-box" class="input-group">
+                        <div class="input-group-btn">
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+<!-- end header part -->
+<!-- Generated by Doxygen 1.8.12 -->
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+var searchBox = new SearchBox("searchBox", "search",false,'Search');
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+<script type="text/javascript" src="menu.js"></script>
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+
+<!-- iframe showing the search results (closed by default) -->
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+
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+<div class="header">
+  <div class="headertitle">
+<div class="title">Build lib &amp; build sample </div>  </div>
+</div><!--header-->
+<div class="contents">
+<div class="toc"><h3>Table of Contents</h3>
+<ul><li class="level1"><a href="#audio_orchestra_build_download">Download:                                          </a><ul><li class="level2"><a href="#audio_orchestra_build_download_repo">need google repo:                                  </a></li>
+<li class="level2"><a href="#audio_orchestra_build_download_lutin">lutin (build-system):                              </a></li>
+<li class="level2"><a href="#audio_orchestra_build_download_dependency">dependency:                                        </a></li>
+<li class="level2"><a href="#audio_orchestra_build_download_sources">sources:                                           </a></li>
+</ul>
+</li>
+<li class="level1"><a href="#audio_orchestra_build_build">Build:                                             </a><ul><li class="level2"><a href="#audio_orchestra_build_build_library">library:                                           </a></li>
+<li class="level2"><a href="#audio_orchestra_build_build_sample">Sample:                                            </a></li>
+</ul>
+</li>
+</ul>
+</div>
+<div class="textblock"><h1><a class="anchor" id="audio_orchestra_build_download"></a>
+Download:                                          </h1>
+<p>ege use some tools to manage source and build it:</p>
+<h2><a class="anchor" id="audio_orchestra_build_download_repo"></a>
+need google repo:                                  </h2>
+<p>see: <a href="http://source.android.com/source/downloading.html#installing-repo">http://source.android.com/source/downloading.html#installing-repo</a></p>
+<p>On all platform: </p><div class="fragment"><div class="line">mkdir ~/.bin</div><div class="line">PATH=~/.bin:$PATH</div><div class="line">curl https://storage.googleapis.com/git-repo-downloads/repo &gt; ~/.bin/repo</div><div class="line">chmod a+x ~/.bin/repo</div></div><!-- fragment --><p>On ubuntu </p><div class="fragment"><div class="line">sudo apt-get install repo</div></div><!-- fragment --><p>On archlinux </p><div class="fragment"><div class="line">sudo pacman -S repo</div></div><!-- fragment --><h2><a class="anchor" id="audio_orchestra_build_download_lutin"></a>
+lutin (build-system):                              </h2>
+<div class="fragment"><div class="line">pip install lutin --user</div><div class="line"># optionnal dependency of lutin (manage image changing size for application release)</div><div class="line">pip install pillow --user</div></div><!-- fragment --><h2><a class="anchor" id="audio_orchestra_build_download_dependency"></a>
+dependency:                                        </h2>
+<div class="fragment"><div class="line">mkdir -p WORKING_DIRECTORY/framework</div><div class="line">cd WORKING_DIRECTORY/framework</div><div class="line">repo init -u git://github.com/atria-soft/manifest.git</div><div class="line">repo sync -j8</div><div class="line">cd ../..</div></div><!-- fragment --><h2><a class="anchor" id="audio_orchestra_build_download_sources"></a>
+sources:                                           </h2>
+<p>They are already download in the repo manifest in:</p>
+<div class="fragment"><div class="line">cd WORKING_DIRECTORY/framework/atria-soft/audio_orchestra</div></div><!-- fragment --><h1><a class="anchor" id="audio_orchestra_build_build"></a>
+Build:                                             </h1>
+<p>you must stay in zour working directory... </p><div class="fragment"><div class="line">cd WORKING_DIRECTORY</div></div><!-- fragment --><h2><a class="anchor" id="audio_orchestra_build_build_library"></a>
+library:                                           </h2>
+<div class="fragment"><div class="line">lutin -mdebug audio-orchestra</div></div><!-- fragment --><h2><a class="anchor" id="audio_orchestra_build_build_sample"></a>
+Sample:                                            </h2>
+<p>No smaple availlable for now ... </p>
+</div></div><!-- contents -->
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+Generated on Mon Oct 24 2016 15:35:48 for Orchestra: audio interface wrapper by &#160;<a href="http://www.doxygen.org/index.html">
+<img class="footer" src="doxygen.png" alt="doxygen"/>
+</a> 1.8.12
+</small></address>
+</body>
+</html>

authors.txt

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

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+  background-color: #e0e0e0;
+}
+
+/* todo lists */
+.todoname,
+.todoname a {
+  font-weight: bold;
+}
+
+/* Class title */
+.summary {
+  margin-top: 25px;
+}
+.page-header {
+  margin: 20px 0px !important;
+}
+.page-header {
+  #display: inline-block;
+}
+.title {
+  text-align: center;
+  color: orange;
+}
+.page-header .pull-right {
+  margin-top: 0.3em;
+  margin-left: 0.5em;
+}
+.page-header .label {
+  font-size: 50%;
+}
+
+#main-nav {
+	display: none;
+}

doc/build.md

@@ -1,83 +0,0 @@
-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

@@ -1,50 +0,0 @@
-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-boot.js

@@ -0,0 +1,264 @@
+$( document ).ready(function() {
+
+	$("div.headertitle").addClass("page-header");
+	$("div.title").addClass("h1");
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+
+	$('div.dyncontent').find('img').each(function(){
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+	$('#navrow1').parent().prepend(nav_container);
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+		left_nav.append(navrow);
+		$('#navrow' + i).remove();
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+		<div id="search-box" class="input-group">\
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+			<ul class="dropdown-menu">\
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+			<button id="search-close" type="button" class="close" aria-label="Close"><span aria-hidden="true">&times;</span></button>\
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+	$(nav_container).append(left_nav);
+	$(nav_container).append(right_nav);
+
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+
+	element_a.click(function(){
+		$('#search-box .dropdown-menu li').removeClass('active');
+		$(this).parent().addClass('active');
+		searchBox.OnSelectItem($('#search-box li a').index(this));
+		searchBox.Search();
+		return false;
+	});
+
+	var element = $('<li></li>').append(element_a);
+	$('#search-box .dropdown-menu').append(element);
+	}
+	$('#MSearchSelectWindow').remove();
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+
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+	$('body').append('<div id="MSearchBox"></div>');
+	$('body').append('<div id="MSearchSelectWindow"></div>');
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+	searchBox.DOMSearchField = function() {
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+	$('#MSearchResultsWindow')
+		.attr('id', 'search-results-window')
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+		.append(
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+			<h3 class="panel-title">Search Results</h3>\
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+			<div class="panel-body"></div>'
+		);
+	$('#search-results-window .panel-body').append(results_iframe);
+
+	searchBox.DOMPopupSearchResultsWindow = function() {
+		return document.getElementById("search-results-window");
+	}
+
+	function update_search_results_window() {
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+		var status = $('#MSearchResults').contents().find('.SRStatus:visible');
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+				break;
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+				$('#search-results-window').addClass('panel-danger');
+				break;
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+			}
+		} else {
+			$('#search-results-window').addClass('panel-success');
+		}
+	}
+	$('#MSearchResults').load(function() {
+	$('#MSearchResults').contents().find('link[href="search.css"]').attr('href','../doxygen.css');
+	$('#MSearchResults').contents().find('head').append(
+		'<link href="../customdoxygen.css" rel="stylesheet" type="text/css">');
+
+	update_search_results_window();
+
+	// detect status changes (only for search with external search backend)
+	var observer = new MutationObserver(function(mutations) {
+		update_search_results_window();
+	});
+	var config = {
+		attributes: true
+	};
+
+	var targets = $('#MSearchResults').contents().find('.SRStatus');
+	for (i = 0; i < targets.length; i++) {
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+
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+	
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+		$(this).find('td > span.warning').parent().addClass('warning');
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+
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+	$("li.navelem").removeClass('navelem');
+	$("a.el").removeClass('el');
+	$("div.ah").removeClass('ah');
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+
+	$('.mdescLeft').each(function(){
+		if($(this).html()=="&nbsp;") {
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+			$(this).remove();
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+			$(this).attr('colspan', 2);
+			$(this).siblings('.memItemRight').remove();
+		}
+	});
+});

doxy_audio-orchestra.py

@@ -1,32 +0,0 @@
-#!/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

dynsections.js

@@ -0,0 +1,97 @@
+function toggleVisibility(linkObj)
+{
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+ var summary = $('#'+base+'-summary');
+ var content = $('#'+base+'-content');
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+ var src=$(trigger).attr('src');
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+   summary.show();
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+   $(trigger).attr('src',src.substring(0,src.length-8)+'closed.png');
+ } else {
+   content.show();
+   summary.hide();
+   $(linkObj).removeClass('closed').addClass('opened');
+   $(trigger).attr('src',src.substring(0,src.length-10)+'open.png');
+ } 
+ return false;
+}
+
+function updateStripes()
+{
+  $('table.directory tr').
+       removeClass('even').filter(':visible:even').addClass('even');
+}
+
+function toggleLevel(level)
+{
+  $('table.directory tr').each(function() {
+    var l = this.id.split('_').length-1;
+    var i = $('#img'+this.id.substring(3));
+    var a = $('#arr'+this.id.substring(3));
+    if (l<level+1) {
+      i.removeClass('iconfopen iconfclosed').addClass('iconfopen');
+      a.html('&#9660;');
+      $(this).show();
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+      a.html('&#9658;');
+      $(this).show();
+    } else {
+      $(this).hide();
+    }
+  });
+  updateStripes();
+}
+
+function toggleFolder(id)
+{
+  // the clicked row
+  var currentRow = $('#row_'+id);
+
+  // all rows after the clicked row
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+
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+
+  // only match elements AFTER this one (can't hide elements before)
+  var childRows = rows.filter(function() { return this.id.match(re); });
+
+  // first row is visible we are HIDING
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+    var currentRowSpans = currentRow.find("span");
+    currentRowSpans.filter(".iconfopen").removeClass("iconfopen").addClass("iconfclosed");
+    currentRowSpans.filter(".arrow").html('&#9658;');
+    rows.filter("[id^=row_"+id+"]").hide(); // hide all children
+  } else { // we are SHOWING
+    // replace right arrow by down arrow for current row
+    var currentRowSpans = currentRow.find("span");
+    currentRowSpans.filter(".iconfclosed").removeClass("iconfclosed").addClass("iconfopen");
+    currentRowSpans.filter(".arrow").html('&#9660;');
+    // replace down arrows by right arrows for child rows
+    var childRowsSpans = childRows.find("span");
+    childRowsSpans.filter(".iconfopen").removeClass("iconfopen").addClass("iconfclosed");
+    childRowsSpans.filter(".arrow").html('&#9658;');
+    childRows.show(); //show all children
+  }
+  updateStripes();
+}
+
+
+function toggleInherit(id)
+{
+  var rows = $('tr.inherit.'+id);
+  var img = $('tr.inherit_header.'+id+' img');
+  var src = $(img).attr('src');
+  if (rows.filter(':first').is(':visible')===true) {
+    rows.css('display','none');
+    $(img).attr('src',src.substring(0,src.length-8)+'closed.png');
+  } else {
+    rows.css('display','table-row'); // using show() causes jump in firefox
+    $(img).attr('src',src.substring(0,src.length-10)+'open.png');
+  }
+}
+

index.html

@@ -0,0 +1,190 @@
+<!-- HTML header for doxygen 1.8.8-->
+<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
+<html xmlns="http://www.w3.org/1999/xhtml">
+    <head>
+        <meta http-equiv="X-UA-Compatible" content="IE=edge">
+        <!-- For Mobile Devices -->
+        <meta name="viewport" content="width=device-width, initial-scale=1">
+        <meta http-equiv="Content-Type" content="text/xhtml;charset=UTF-8"/>
+        <meta name="generator" content="Doxygen 1.8.12"/>
+        <script type="text/javascript" src="https://code.jquery.com/jquery-2.1.1.min.js"></script>
+        <title>Orchestra: audio interface wrapper: AUDIO-ORCHESTRA library</title>
+        <!--<link href="tabs.css" rel="stylesheet" type="text/css"/>-->
+        <script type="text/javascript" src="dynsections.js"></script>
+        <link href="search/search.css" rel="stylesheet" type="text/css"/>
+<script type="text/javascript" src="search/searchdata.js"></script>
+<script type="text/javascript" src="search/search.js"></script>
+        <link href="doxygen.css" rel="stylesheet" type="text/css" />
+        <link href="customdoxygen.css" rel="stylesheet" type="text/css"/>
+        <link rel="stylesheet" href="https://maxcdn.bootstrapcdn.com/bootstrap/3.3.1/css/bootstrap.min.css">
+        <script src="https://maxcdn.bootstrapcdn.com/bootstrap/3.3.1/js/bootstrap.min.js"></script>
+        <script type="text/javascript" src="doxy-boot.js"></script>
+    </head>
+    <body>
+        <nav class="navbar navbar-default navbar-fixed-top" role="navigation">
+            <div class="container">
+                <div class="navbar-header">
+                    <a class="navbar-brand">Orchestra: audio interface wrapper 0.4.0</a>
+                </div>
+                <div id="navbar" class="navbar-collapse collapse">
+                    <ul class="nav navbar-nav">
+                        <li><a href="index.html">Main&nbsp;Page</a></li>
+                        <li><a href="pages.html">Related&nbsp;Pages</a></li>
+                        <li><a href="namespaces.html">Namespaces</a></li>
+                        <li><a href="annotated.html">Classes</a></li>
+                        <li><a href="files.html">Files</a></li>
+                    </ul>
+                    <ul class="nav navbar-nav navbar-right">
+                        <li class="dropdown">
+                            <a href="#" class="dropdown-toggle" data-toggle="dropdown" role="button" aria-haspopup="true" aria-expanded="false">
+                                Link-libs<span class="caret"></span>
+                            </a>
+                            <ul class="dropdown-menu">
+                                <li><a href="http://HeeroYui.github.io/lutin">lutin</a></li>
+                                <li><a href="http://atria-soft.github.io/ewol">ewol</a></li>
+                                <li><a href="http://atria-soft.github.io/echrono">echrono</a></li>
+                                <li><a href="http://atria-soft.github.io/etk">etk</a></li>
+                                <li><a href="http://atria-soft.github.io/ejson">ejson</a></li>
+                                <li><a href="http://atria-soft.github.io/exml">exml</a></li>
+                                <li><a href="http://atria-soft.github.io/esvg">esvg</a></li>
+                                <li><a href="http://atria-soft.github.io/egami">egami</a></li>
+                                <li><a href="http://atria-soft.github.io/gale">gale</a></li>
+                                <li><a href="http://atria-soft.github.io/ege">ege</a></li>
+                                <li><a href="http://atria-soft.github.io/elog">elog</a></li>
+                                <li><a href="http://atria-soft.github.io/ememory">ememory</a></li>
+                                <li><a href="http://atria-soft.github.io/enet">enet</a></li>
+                                <li><a href="http://atria-soft.github.io/eproperty">eproperty</a></li>
+                                <li><a href="http://atria-soft.github.io/esignal">esignal</a></li>
+                                <li><a href="http://atria-soft.github.io/etranslate">etranslate</a></li>
+                                <li><a href="http://atria-soft.github.io/zeus">zeus</a></li>
+                                <li><a href="http://musicdsp.github.io/audio-ess">audio-ess</a></li>
+                                <li><a href="http://musicdsp.github.io/audio">audio</a></li>
+                                <li><a href="http://musicdsp.github.io/audio-drain">audio-drain</a></li>
+                                <li><a href="http://musicdsp.github.io/audio-orchestra">audio-orchestra</a></li>
+                                <li><a href="http://musicdsp.github.io/audio-river">audio-river</a></li>
+                            </ul>
+                        </li>
+                    </ul>
+                    <div id="search-box" class="input-group">
+                        <div class="input-group-btn">
+                            <button aria-expanded="false" type="button" class="btn btn-default dropdown-toggle" data-toggle="dropdown">
+                                <span class="glyphicon glyphicon-search"></span>
+                                <span class="caret"></span>
+                            </button>
+                            <ul class="dropdown-menu">
+                                <li><a href="#">All</a></li>
+                                <li><a href="#">Classes</a></li>
+                                <li><a href="#">Namespaces</a></li>
+                                <li><a href="#">Files</a></li>
+                                <li><a href="#">Functions</a></li>
+                                <li><a href="#">Variables</a></li>
+                                <li><a href="#">Typedefs</a></li>
+                                <li><a href="#">Enumerations</a></li>
+                                <li><a href="#">Enumerator</a></li>
+                                <li><a href="#">Friends</a></li>
+                                <li><a href="#">Macros</a></li>
+                                <li><a href="#">Pages</a></li>
+                            </ul>
+                        </div>
+                        <button id="search-close" type="button" class="close" aria-label="Close">
+                            <span aria-hidden="true"></span>
+                        </button>
+                        <input id="search-field" class="form-control" accesskey="S" onkeydown="searchBox.OnSearchFieldChange(event);" placeholder="Search ..." type="text">
+                    </div>
+                </div><!--/.nav-collapse -->
+            </div>
+        </nav>
+        <div id="top"><!-- do not remove this div, it is closed by doxygen! -->
+            <div class="content" id="content">
+                <div class="container">
+                    <div class="row">
+                        <div class="col-sm-12 panel panel-default" style="padding-bottom: 15px;">
+                            <div style="margin-bottom: 15px;margin-top: 60px;">
+<!-- end header part -->
+<!-- Generated by Doxygen 1.8.12 -->
+<script type="text/javascript">
+var searchBox = new SearchBox("searchBox", "search",false,'Search');
+</script>
+<script type="text/javascript" src="menudata.js"></script>
+<script type="text/javascript" src="menu.js"></script>
+<script type="text/javascript">
+$(function() {
+  initMenu('',true,false,'search.php','Search');
+  $(document).ready(function() { init_search(); });
+});
+</script>
+<div id="main-nav"></div>
+</div><!-- top -->
+<!-- window showing the filter options -->
+<div id="MSearchSelectWindow"
+     onmouseover="return searchBox.OnSearchSelectShow()"
+     onmouseout="return searchBox.OnSearchSelectHide()"
+     onkeydown="return searchBox.OnSearchSelectKey(event)">
+</div>
+
+<!-- iframe showing the search results (closed by default) -->
+<div id="MSearchResultsWindow">
+<iframe src="javascript:void(0)" frameborder="0" 
+        name="MSearchResults" id="MSearchResults">
+</iframe>
+</div>
+
+<div class="header">
+  <div class="headertitle">
+<div class="title">AUDIO-ORCHESTRA library </div>  </div>
+</div><!--header-->
+<div class="contents">
+<div class="toc"><h3>Table of Contents</h3>
+<ul><li class="level1"><a href="#audio_orchestra_mainpage_what">What is AUDIO-ORCHESTRA:                               </a><ul><li class="level2"><a href="#audio_orchestra_mainpage_what_it_does">What it does:                               </a></li>
+</ul>
+</li>
+<li class="level1"><a href="#audio_orchestra_mainpage_language">What languages are supported?                    </a></li>
+<li class="level1"><a href="#audio_orchestra_mainpage_license_restriction">Are there any licensing restrictions?            </a></li>
+<li class="level1"><a href="#audio_orchestra_mainpage_license">License (MIT)                             </a></li>
+<li class="level1"><a href="#audio_orchestra_mainpage_sub_page">Other pages                              </a></li>
+</ul>
+</div>
+<div class="textblock"><h1><a class="anchor" id="audio_orchestra_mainpage_what"></a>
+What is AUDIO-ORCHESTRA:                               </h1>
+<p>AUDIO-ORCHESTRA, is a fork of RTAudio lib (with port for Android, and IOs).</p>
+<p>This is a cross API of Audio wrapping the Hardware.</p>
+<p>This library is not friendy usable, use audio-river to have a correct and simple multiple-flow API</p>
+<h2><a class="anchor" id="audio_orchestra_mainpage_what_it_does"></a>
+What it does:                               </h2>
+<ul>
+<li>Open a strem audio in Input or output</li>
+<li>synchronise 2 flow</li>
+<li>open on some platform: Android, Linux, MacOs, Ios, Windows</li>
+</ul>
+<p>AUDIO-ORCHESTRA is dependent of the STL (compatible with MacOs stl (CXX))</p>
+<h1><a class="anchor" id="audio_orchestra_mainpage_language"></a>
+What languages are supported?                    </h1>
+<p>AUDIO-ORCHESTRA is written in C++.</p>
+<h1><a class="anchor" id="audio_orchestra_mainpage_license_restriction"></a>
+Are there any licensing restrictions?            </h1>
+<p>AUDIO-ORCHESTRA is <b>FREE software</b> and <em>all sub-library are FREE and staticly linkable !!!</em></p>
+<h1><a class="anchor" id="audio_orchestra_mainpage_license"></a>
+License (MIT)                             </h1>
+<p>Copyright AUDIO-ORCHESTRA Edouard DUPIN</p>
+<p>MIT ...</p>
+<h1><a class="anchor" id="audio_orchestra_mainpage_sub_page"></a>
+Other pages                              </h1>
+<ul>
+<li><a class="el" href="audio_orchestra_build.html">Build lib &amp; build sample</a></li>
+<li><a href="http://atria-soft.github.io/ewol/ewol_coding_style.html"><b>ewol coding style</b></a> </li>
+</ul>
+</div></div><!-- contents -->
+<!-- HTML footer for doxygen 1.8.8-->
+<!-- start footer part -->
+</div>
+</div>
+</div>
+</div>
+</div>
+<hr class="footer"/><address class="footer"><small>
+Generated on Mon Oct 24 2016 15:35:48 for Orchestra: audio interface wrapper by &#160;<a href="http://www.doxygen.org/index.html">
+<img class="footer" src="doxygen.png" alt="doxygen"/>
+</a> 1.8.12
+</small></address>
+</body>
+</html>

lutin_audio-orchestra.py

@@ -1,182 +0,0 @@
-#!/usr/bin/python
-import lutin.tools as tools
-import realog.debug as debug
-
-
-def get_type():
-	return "LIBRARY"
-
-def get_desc():
-	return "Generic wrapper on all audio interface"
-
-def get_licence():
-	return "APACHE-2"
-
-def get_compagny_type():
-	return "com"
-
-def get_compagny_name():
-	return "atria-soft"
-
-def get_maintainer():
-	return "authors.txt"
-
-def get_version():
-	return "version.txt"
-
-def configure(target, my_module):
-	my_module.add_src_file([
-		'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'
-		])
-	my_module.add_header_file([
-		'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_depend([
-	    'audio',
-	    'etk'
-	    ])
-	# add all the time the dummy interface
-	my_module.add_flag('c++', ['-DORCHESTRA_BUILD_DUMMY'], export=True)
-	# TODO : Add a FILE interface:
-	
-	if "Windows" in target.get_type():
-		my_module.add_src_file([
-		    'audio/orchestra/api/Asio.cpp',
-		    'audio/orchestra/api/Ds.cpp',
-		    ])
-		# load optionnal API:
-		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_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'
-		    ])
-		# MacOsX core
-		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_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 interfacef
-		my_module.add_src_file('org.musicdsp.orchestra.OrchestraConstants.javah')
-		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')
-		# add tre creator of the basic java class ...
-		target.add_action("BINARY", 11, "audio-orchestra-out-wrapper", tool_generate_add_java_section_in_class)
-	else:
-		debug.warning("unknow target for audio_orchestra : " + target.name);
-	
-	my_module.add_path(".")
-	
-	return True
-
-
-
-##################################################################
-##
-## Android specific section
-##
-##################################################################
-def tool_generate_add_java_section_in_class(target, module, package_name):
-	module.add_pkg("GENERATE_SECTION__IMPORT", [
-		"import org.musicdsp.orchestra.OrchestraManager;"
-		])
-	module.add_pkg("GENERATE_SECTION__DECLARE", [
-		"private OrchestraManager m_audioManagerHandle;"
-		])
-	module.add_pkg("GENERATE_SECTION__CONSTRUCTOR", [
-		"// load audio maneger if it does not work, it is not critical ...",
-		"try {",
-		"	m_audioManagerHandle = new OrchestraManager();",
-		"} catch (RuntimeException e) {",
-		"	Log.e(\"" + package_name + "\", \"Can not load Audio interface (maybe not really needed) :\" + e);",
-		"}"
-		])
-	module.add_pkg("GENERATE_SECTION__ON_CREATE", [
-		"if (m_audioManagerHandle != null) {",
-		"	m_audioManagerHandle.onCreate();",
-		"}"
-		])
-	module.add_pkg("GENERATE_SECTION__ON_START", [
-		"if (m_audioManagerHandle != null) {",
-		"	m_audioManagerHandle.onStart();",
-		"}"
-		])
-	module.add_pkg("GENERATE_SECTION__ON_RESTART", [
-		"if (m_audioManagerHandle != null) {",
-		"	m_audioManagerHandle.onRestart();",
-		"}"
-		])
-	module.add_pkg("GENERATE_SECTION__ON_RESUME", [
-		"if (m_audioManagerHandle != null) {",
-		"	m_audioManagerHandle.onResume();",
-		"}"
-		])
-	module.add_pkg("GENERATE_SECTION__ON_PAUSE", [
-		"if (m_audioManagerHandle != null) {",
-		"	m_audioManagerHandle.onPause();",
-		"}"
-		])
-	module.add_pkg("GENERATE_SECTION__ON_STOP", [
-		"if (m_audioManagerHandle != null) {",
-		"	m_audioManagerHandle.onStop();",
-		"}"
-		])
-	module.add_pkg("GENERATE_SECTION__ON_DESTROY", [
-		"// Destroy the AdView.",
-		"if (m_audioManagerHandle != null) {",
-		"	m_audioManagerHandle.onDestroy();",
-		"}"
-		])
-	
-
-
-
-

menu.js

@@ -0,0 +1,26 @@
+function initMenu(relPath,searchEnabled,serverSide,searchPage,search) {
+  function makeTree(data,relPath) {
+    var result='';
+    if ('children' in data) {
+      result+='<ul>';
+      for (var i in data.children) {
+        result+='<li><a href="'+relPath+data.children[i].url+'">'+
+                                data.children[i].text+'</a>'+
+                                makeTree(data.children[i],relPath)+'</li>';
+      }
+      result+='</ul>';
+    }
+    return result;
+  }
+
+  $('#main-nav').append(makeTree(menudata,relPath));
+  $('#main-nav').children(':first').addClass('sm sm-dox').attr('id','main-menu');
+  if (searchEnabled) {
+    if (serverSide) {
+      $('#main-menu').append('<li style="float:right"><div id="MSearchBox" class="MSearchBoxInactive"><div class="left"><form id="FSearchBox" action="'+searchPage+'" method="get"><img id="MSearchSelect" src="'+relPath+'search/mag.png" alt=""/><input type="text" id="MSearchField" name="query" value="'+search+'" size="20" accesskey="S" onfocus="searchBox.OnSearchFieldFocus(true)" onblur="searchBox.OnSearchFieldFocus(false)"></form></div><div class="right"></div></div></li>');
+    } else {
+      $('#main-menu').append('<li style="float:right"><div id="MSearchBox" class="MSearchBoxInactive"><span class="left"><img id="MSearchSelect" src="'+relPath+'search/mag_sel.png" onmouseover="return searchBox.OnSearchSelectShow()" onmouseout="return searchBox.OnSearchSelectHide()" alt=""/><input type="text" id="MSearchField" value="'+search+'" accesskey="S" onfocus="searchBox.OnSearchFieldFocus(true)" onblur="searchBox.OnSearchFieldFocus(false)" onkeyup="searchBox.OnSearchFieldChange(event)"/></span><span class="right"><a id="MSearchClose" href="javascript:searchBox.CloseResultsWindow()"><img id="MSearchCloseImg" border="0" src="'+relPath+'search/close.png" alt=""/></a></span></div></li>');
+    }
+  }
+  $('#main-menu').smartmenus();
+}

menudata.js

@@ -0,0 +1,3 @@
+var menudata={children:[
+{text:'Main Page',url:'index.html'},
+{text:'Related Pages',url:'pages.html'}]}

pages.html

@@ -0,0 +1,156 @@
+<!-- HTML header for doxygen 1.8.8-->
+<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
+<html xmlns="http://www.w3.org/1999/xhtml">
+    <head>
+        <meta http-equiv="X-UA-Compatible" content="IE=edge">
+        <!-- For Mobile Devices -->
+        <meta name="viewport" content="width=device-width, initial-scale=1">
+        <meta http-equiv="Content-Type" content="text/xhtml;charset=UTF-8"/>
+        <meta name="generator" content="Doxygen 1.8.12"/>
+        <script type="text/javascript" src="https://code.jquery.com/jquery-2.1.1.min.js"></script>
+        <title>Orchestra: audio interface wrapper: Related Pages</title>
+        <!--<link href="tabs.css" rel="stylesheet" type="text/css"/>-->
+        <script type="text/javascript" src="dynsections.js"></script>
+        <link href="search/search.css" rel="stylesheet" type="text/css"/>
+<script type="text/javascript" src="search/searchdata.js"></script>
+<script type="text/javascript" src="search/search.js"></script>
+        <link href="doxygen.css" rel="stylesheet" type="text/css" />
+        <link href="customdoxygen.css" rel="stylesheet" type="text/css"/>
+        <link rel="stylesheet" href="https://maxcdn.bootstrapcdn.com/bootstrap/3.3.1/css/bootstrap.min.css">
+        <script src="https://maxcdn.bootstrapcdn.com/bootstrap/3.3.1/js/bootstrap.min.js"></script>
+        <script type="text/javascript" src="doxy-boot.js"></script>
+    </head>
+    <body>
+        <nav class="navbar navbar-default navbar-fixed-top" role="navigation">
+            <div class="container">
+                <div class="navbar-header">
+                    <a class="navbar-brand">Orchestra: audio interface wrapper 0.4.0</a>
+                </div>
+                <div id="navbar" class="navbar-collapse collapse">
+                    <ul class="nav navbar-nav">
+                        <li><a href="index.html">Main&nbsp;Page</a></li>
+                        <li><a href="pages.html">Related&nbsp;Pages</a></li>
+                        <li><a href="namespaces.html">Namespaces</a></li>
+                        <li><a href="annotated.html">Classes</a></li>
+                        <li><a href="files.html">Files</a></li>
+                    </ul>
+                    <ul class="nav navbar-nav navbar-right">
+                        <li class="dropdown">
+                            <a href="#" class="dropdown-toggle" data-toggle="dropdown" role="button" aria-haspopup="true" aria-expanded="false">
+                                Link-libs<span class="caret"></span>
+                            </a>
+                            <ul class="dropdown-menu">
+                                <li><a href="http://HeeroYui.github.io/lutin">lutin</a></li>
+                                <li><a href="http://atria-soft.github.io/ewol">ewol</a></li>
+                                <li><a href="http://atria-soft.github.io/echrono">echrono</a></li>
+                                <li><a href="http://atria-soft.github.io/etk">etk</a></li>
+                                <li><a href="http://atria-soft.github.io/ejson">ejson</a></li>
+                                <li><a href="http://atria-soft.github.io/exml">exml</a></li>
+                                <li><a href="http://atria-soft.github.io/esvg">esvg</a></li>
+                                <li><a href="http://atria-soft.github.io/egami">egami</a></li>
+                                <li><a href="http://atria-soft.github.io/gale">gale</a></li>
+                                <li><a href="http://atria-soft.github.io/ege">ege</a></li>
+                                <li><a href="http://atria-soft.github.io/elog">elog</a></li>
+                                <li><a href="http://atria-soft.github.io/ememory">ememory</a></li>
+                                <li><a href="http://atria-soft.github.io/enet">enet</a></li>
+                                <li><a href="http://atria-soft.github.io/eproperty">eproperty</a></li>
+                                <li><a href="http://atria-soft.github.io/esignal">esignal</a></li>
+                                <li><a href="http://atria-soft.github.io/etranslate">etranslate</a></li>
+                                <li><a href="http://atria-soft.github.io/zeus">zeus</a></li>
+                                <li><a href="http://musicdsp.github.io/audio-ess">audio-ess</a></li>
+                                <li><a href="http://musicdsp.github.io/audio">audio</a></li>
+                                <li><a href="http://musicdsp.github.io/audio-drain">audio-drain</a></li>
+                                <li><a href="http://musicdsp.github.io/audio-orchestra">audio-orchestra</a></li>
+                                <li><a href="http://musicdsp.github.io/audio-river">audio-river</a></li>
+                            </ul>
+                        </li>
+                    </ul>
+                    <div id="search-box" class="input-group">
+                        <div class="input-group-btn">
+                            <button aria-expanded="false" type="button" class="btn btn-default dropdown-toggle" data-toggle="dropdown">
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+                                <li><a href="#">Typedefs</a></li>
+                                <li><a href="#">Enumerations</a></li>
+                                <li><a href="#">Enumerator</a></li>
+                                <li><a href="#">Friends</a></li>
+                                <li><a href="#">Macros</a></li>
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+                            </ul>
+                        </div>
+                        <button id="search-close" type="button" class="close" aria-label="Close">
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