From 7b0316a8aa0f99d5263cacab48574798c6a345ff Mon Sep 17 00:00:00 2001
From: Edouard DUPIN <yui.heero@gmail.com>
Date: Sun, 14 Jun 2015 18:32:08 +0200
Subject: [PATCH] [DEV] rework continue (better integration of pulseaudio and
 low level devices

---
 audio/orchestra/DeviceInfo.cpp          |  24 +-
 audio/orchestra/DeviceInfo.h            |  25 +-
 audio/orchestra/Interface.cpp           |  52 +-
 audio/orchestra/Interface.h             |  66 +--
 audio/orchestra/api/Alsa.cpp            | 152 ++----
 audio/orchestra/api/Alsa.h              |   4 +-
 audio/orchestra/api/Ds.cpp              | 607 +++++++++++-------------
 audio/orchestra/api/Ds.h                |   2 -
 audio/orchestra/api/Jack.cpp            |  73 ++-
 audio/orchestra/api/Pulse.cpp           |  25 +-
 audio/orchestra/api/PulseDeviceList.cpp | 262 ++++++++--
 audio/orchestra/api/PulseDeviceList.h   |  30 +-
 12 files changed, 666 insertions(+), 656 deletions(-)

diff --git a/audio/orchestra/DeviceInfo.cpp b/audio/orchestra/DeviceInfo.cpp
index 9cc645e..ca58a14 100644
--- a/audio/orchestra/DeviceInfo.cpp
+++ b/audio/orchestra/DeviceInfo.cpp
@@ -19,26 +19,22 @@ void audio::orchestra::DeviceInfo::display(int32_t _tabNumber) const {
 	for (int32_t iii=0; iii<_tabNumber; ++iii) {
 		space += "    ";
 	}
-	ATA_PRINT(space + "probe=" << probed);
+	ATA_PRINT(space + "mode=" << (input==true?"input":"output"));
 	ATA_PRINT(space + "name=" << name);
-	ATA_PRINT(space + "outputChannels=" << outputChannels);
-	ATA_PRINT(space + "inputChannels=" << inputChannels);
-	ATA_PRINT(space + "duplexChannels=" << duplexChannels);
-	ATA_PRINT(space + "isDefaultOutput=" << (isDefaultOutput==true?"true":"false"));
-	ATA_PRINT(space + "isDefaultInput=" << (isDefaultInput==true?"true":"false"));
-	ATA_PRINT(space + "rates=" << sampleRates);
-	ATA_PRINT(space + "native Format: " << nativeFormats);
+	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"));
 }
 
+
 std::ostream& audio::orchestra::operator <<(std::ostream& _os, const audio::orchestra::DeviceInfo& _obj) {
 	_os << "{";
-	_os << "probe=" << _obj.probed << ", ";
 	_os << "name=" << _obj.name << ", ";
-	_os << "outputChannels=" << _obj.outputChannels << ", ";
-	_os << "inputChannels=" << _obj.inputChannels << ", ";
-	_os << "duplexChannels=" << _obj.duplexChannels << ", ";
-	_os << "isDefaultOutput=" << _obj.isDefaultOutput << ", ";
-	_os << "isDefaultInput=" << _obj.isDefaultInput << ", ";
+	_os << "description=" << _obj.desc << ", ";
+	_os << "channels=" << _obj.channels << ", ";
+	_os << "default=" << _obj.isDefault << ", ";
 	_os << "rates=" << _obj.sampleRates << ", ";
 	_os << "native Format: " << _obj.nativeFormats;
 	_os << "}";
diff --git a/audio/orchestra/DeviceInfo.h b/audio/orchestra/DeviceInfo.h
index 8b0d9ba..ec8875d 100644
--- a/audio/orchestra/DeviceInfo.h
+++ b/audio/orchestra/DeviceInfo.h
@@ -9,6 +9,7 @@
 #define __AUDIO_ORCHESTRA_DEVICE_INFO_H__
 
 #include <audio/format.h>
+#include <audio/channel.h>
 
 
 namespace audio {
@@ -18,24 +19,22 @@ namespace audio {
 		 */
 		class DeviceInfo {
 			public:
-				bool probed; //!< true if the device capabilities were successfully probed.
+				bool input; //!< true if the device in an input; false: output.
 				std::string name; //!< Character string device identifier.
-				uint32_t outputChannels; //!< Maximum output channels supported by device.
-				uint32_t inputChannels; //!< Maximum input channels supported by device.
-				uint32_t duplexChannels; //!< Maximum simultaneous input/output channels supported by device.
-				bool isDefaultOutput; //!< true if this is the default output device.
-				bool isDefaultInput; //!< true if this is the default input device.
+				std::string desc; //!< description of the device
+				std::vector<audio::channel> channels; //!< Channels interfaces.
 				std::vector<uint32_t> sampleRates; //!< Supported sample rates (queried from list of standard rates).
 				std::vector<audio::format> nativeFormats; //!< Bit mask of supported data formats.
+				bool isDefault; //! is default input/output
 				// Default constructor.
 				DeviceInfo() :
-				  probed(false),
-				  outputChannels(0),
-				  inputChannels(0),
-				  duplexChannels(0),
-				  isDefaultOutput(false),
-				  isDefaultInput(false),
-				  nativeFormats() {}
+				  input(false),
+				  name(),
+				  desc(),
+				  channels(),
+				  sampleRates(),
+				  nativeFormats(),
+				  isDefault(false) {}
 				void display(int32_t _tabNumber = 1) const;
 		};
 		std::ostream& operator <<(std::ostream& _os, const audio::orchestra::DeviceInfo& _obj);
diff --git a/audio/orchestra/Interface.cpp b/audio/orchestra/Interface.cpp
index d57158a..17f58f1 100644
--- a/audio/orchestra/Interface.cpp
+++ b/audio/orchestra/Interface.cpp
@@ -26,14 +26,14 @@ std::vector<std::string> audio::orchestra::Interface::getListApi() {
 
 
 void audio::orchestra::Interface::openApi(const std::string& _api) {
-	delete m_rtapi;
-	m_rtapi = nullptr;
+	delete m_api;
+	m_api = nullptr;
 	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_rtapi = m_apiAvaillable[iii].second();
-			if (m_rtapi != nullptr) {
+			m_api = m_apiAvaillable[iii].second();
+			if (m_api != nullptr) {
 				return;
 			}
 		}
@@ -44,7 +44,7 @@ void audio::orchestra::Interface::openApi(const std::string& _api) {
 
 
 audio::orchestra::Interface::Interface() :
-  m_rtapi(nullptr) {
+  m_api(nullptr) {
 	ATA_DEBUG("Add interface:");
 #if defined(ORCHESTRA_BUILD_JACK)
 	addInterface(audio::orchestra::type_jack, audio::orchestra::api::Jack::create);
@@ -84,18 +84,18 @@ void audio::orchestra::Interface::addInterface(const std::string& _api, Api* (*_
 
 enum audio::orchestra::error audio::orchestra::Interface::clear() {
 	ATA_INFO("Clear API ...");
-	if (m_rtapi == nullptr) {
+	if (m_api == nullptr) {
 		ATA_WARNING("Interface NOT started!");
 		return audio::orchestra::error_none;
 	}
-	delete m_rtapi;
-	m_rtapi = nullptr;
+	delete m_api;
+	m_api = nullptr;
 	return audio::orchestra::error_none;
 }
 
 enum audio::orchestra::error audio::orchestra::Interface::instanciate(const std::string& _api) {
 	ATA_INFO("Instanciate API ...");
-	if (m_rtapi != nullptr) {
+	if (m_api != nullptr) {
 		ATA_WARNING("Interface already started!");
 		return audio::orchestra::error_none;
 	}
@@ -103,15 +103,15 @@ enum audio::orchestra::error audio::orchestra::Interface::instanciate(const std:
 		ATA_INFO("API specified : " << _api);
 		// Attempt to open the specified API.
 		openApi(_api);
-		if (m_rtapi != nullptr) {
-			if (m_rtapi->getDeviceCount() != 0) {
+		if (m_api != nullptr) {
+			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("RtAudio: no compiled support for specified API argument!");
+		ATA_ERROR("API NOT Supported '" << _api << "' not in " << getListApi());
 		return audio::orchestra::error_fail;
 	}
 	ATA_INFO("Auto choice API :");
@@ -122,26 +122,26 @@ enum audio::orchestra::error audio::orchestra::Interface::instanciate(const std:
 	for (size_t iii=0; iii<apis.size(); ++iii) {
 		ATA_INFO("try open ...");
 		openApi(apis[iii]);
-		if(m_rtapi == nullptr) {
+		if(m_api == nullptr) {
 			ATA_ERROR("    ==> can not create ...");
 			continue;
 		}
-		if (m_rtapi->getDeviceCount() != 0) {
+		if (m_api->getDeviceCount() != 0) {
 			ATA_INFO("    ==> api open");
 			break;
 		}
 	}
-	if (m_rtapi != nullptr) {
+	if (m_api != nullptr) {
 		return audio::orchestra::error_none;
 	}
-	ATA_ERROR("RtAudio: no compiled API support found ... critical error!!");
+	ATA_ERROR("API NOT Supported '" << _api << "' not in " << getListApi());
 	return audio::orchestra::error_fail;
 }
 
 audio::orchestra::Interface::~Interface() {
 	ATA_INFO("Remove interface");
-	delete m_rtapi;
-	m_rtapi = nullptr;
+	delete m_api;
+	m_api = nullptr;
 }
 
 enum audio::orchestra::error audio::orchestra::Interface::openStream(audio::orchestra::StreamParameters* _outputParameters,
@@ -151,10 +151,10 @@ enum audio::orchestra::error audio::orchestra::Interface::openStream(audio::orch
                                                                      uint32_t* _bufferFrames,
                                                                      audio::orchestra::AirTAudioCallback _callback,
                                                                      const audio::orchestra::StreamOptions& _options) {
-	if (m_rtapi == nullptr) {
+	if (m_api == nullptr) {
 		return audio::orchestra::error_inputNull;
 	}
-	return m_rtapi->openStream(_outputParameters,
+	return m_api->openStream(_outputParameters,
 	                           _inputParameters,
 	                           _format,
 	                           _sampleRate,
@@ -164,22 +164,22 @@ enum audio::orchestra::error audio::orchestra::Interface::openStream(audio::orch
 }
 
 bool audio::orchestra::Interface::isMasterOf(audio::orchestra::Interface& _interface) {
-	if (m_rtapi == nullptr) {
+	if (m_api == nullptr) {
 		ATA_ERROR("Current Master API is nullptr ...");
 		return false;
 	}
-	if (_interface.m_rtapi == nullptr) {
+	if (_interface.m_api == nullptr) {
 		ATA_ERROR("Current Slave API is nullptr ...");
 		return false;
 	}
-	if (m_rtapi->getCurrentApi() != _interface.m_rtapi->getCurrentApi()) {
+	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_rtapi->getCurrentApi() != audio::orchestra::type_alsa) {
-		ATA_ERROR("Link 2 device together work only if the interafec is ?");// << audio::orchestra::type_alsa << " not for " << m_rtapi->getCurrentApi());
+	if (m_api->getCurrentApi() != audio::orchestra::type_alsa) {
+		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_rtapi->isMasterOf(_interface.m_rtapi);
+	return m_api->isMasterOf(_interface.m_api);
 }
 
diff --git a/audio/orchestra/Interface.h b/audio/orchestra/Interface.h
index 6a7f0eb..8f5c4e5 100644
--- a/audio/orchestra/Interface.h
+++ b/audio/orchestra/Interface.h
@@ -40,14 +40,14 @@ namespace audio {
 			protected:
 				std::vector<std::pair<std::string, Api* (*)()> > m_apiAvaillable;
 			protected:
-				audio::orchestra::Api *m_rtapi;
+				audio::orchestra::Api *m_api;
 			public:
 				void setName(const std::string& _name) {
-					if (m_rtapi == nullptr) {
+					if (m_api == nullptr) {
 						
 						return;
 					}
-					m_rtapi->setName(_name);
+					m_api->setName(_name);
 				}
 				/**
 				 * @brief Get the list of all availlable API in the system.
@@ -84,10 +84,10 @@ namespace audio {
 				 * @return the audio API specifier for the current instance of airtaudio.
 				 */
 				const std::string& getCurrentApi() {
-					if (m_rtapi == nullptr) {
+					if (m_api == nullptr) {
 						return audio::orchestra::type_undefined;
 					}
-					return m_rtapi->getCurrentApi();
+					return m_api->getCurrentApi();
 				}
 				/**
 				 * @brief A public function that queries for the number of audio devices available.
@@ -97,10 +97,10 @@ namespace audio {
 				 * a system error occurs during processing, a warning will be issued. 
 				 */
 				uint32_t getDeviceCount() {
-					if (m_rtapi == nullptr) {
+					if (m_api == nullptr) {
 						return 0;
 					}
-					return m_rtapi->getDeviceCount();
+					return m_api->getDeviceCount();
 				}
 				/**
 				 * @brief Any device integer between 0 and getDeviceCount() - 1 is valid.
@@ -114,17 +114,17 @@ namespace audio {
 				 * @return An audio::orchestra::DeviceInfo structure for a specified device number.
 				 */
 				audio::orchestra::DeviceInfo getDeviceInfo(uint32_t _device) {
-					if (m_rtapi == nullptr) {
+					if (m_api == nullptr) {
 						return audio::orchestra::DeviceInfo();
 					}
-					return m_rtapi->getDeviceInfo(_device);
+					return m_api->getDeviceInfo(_device);
 				}
 				audio::orchestra::DeviceInfo getDeviceInfo(const std::string& _deviceName) {
-					if (m_rtapi == nullptr) {
+					if (m_api == nullptr) {
 						return audio::orchestra::DeviceInfo();
 					}
 					audio::orchestra::DeviceInfo info;
-					m_rtapi->getNamedDeviceInfo(_deviceName, info);
+					m_api->getNamedDeviceInfo(_deviceName, info);
 					return info;
 				}
 				/**
@@ -137,10 +137,10 @@ namespace audio {
 				 * before attempting to open a stream.
 				 */
 				uint32_t getDefaultOutputDevice() {
-					if (m_rtapi == nullptr) {
+					if (m_api == nullptr) {
 						return 0;
 					}
-					return m_rtapi->getDefaultOutputDevice();
+					return m_api->getDefaultOutputDevice();
 				}
 				/**
 				 * @brief A function that returns the index of the default input device.
@@ -152,10 +152,10 @@ namespace audio {
 				 * before attempting to open a stream.
 				 */
 				uint32_t getDefaultInputDevice() {
-					if (m_rtapi == nullptr) {
+					if (m_api == nullptr) {
 						return 0;
 					}
-					return m_rtapi->getDefaultInputDevice();
+					return m_api->getDefaultInputDevice();
 				}
 				/**
 				 * @brief A public function for opening a stream with the specified parameters.
@@ -210,10 +210,10 @@ namespace audio {
 				 * returns (no exception is thrown).
 				 */
 				enum audio::orchestra::error closeStream() {
-					if (m_rtapi == nullptr) {
+					if (m_api == nullptr) {
 						return audio::orchestra::error_inputNull;
 					}
-					return m_rtapi->closeStream();
+					return m_api->closeStream();
 				}
 				/**
 				 * @brief A function that starts a stream.
@@ -224,10 +224,10 @@ namespace audio {
 				 * running.
 				 */
 				enum audio::orchestra::error startStream() {
-					if (m_rtapi == nullptr) {
+					if (m_api == nullptr) {
 						return audio::orchestra::error_inputNull;
 					}
-					return m_rtapi->startStream();
+					return m_api->startStream();
 				}
 				/**
 				 * @brief Stop a stream, allowing any samples remaining in the output queue to be played.
@@ -238,10 +238,10 @@ namespace audio {
 				 * stopped.
 				*/
 				enum audio::orchestra::error stopStream() {
-					if (m_rtapi == nullptr) {
+					if (m_api == nullptr) {
 						return audio::orchestra::error_inputNull;
 					}
-					return m_rtapi->stopStream();
+					return m_api->stopStream();
 				}
 				/**
 				 * @brief Stop a stream, discarding any samples remaining in the input/output queue.
@@ -251,38 +251,38 @@ namespace audio {
 				 * stopped.
 				 */
 				enum audio::orchestra::error abortStream() {
-					if (m_rtapi == nullptr) {
+					if (m_api == nullptr) {
 						return audio::orchestra::error_inputNull;
 					}
-					return m_rtapi->abortStream();
+					return m_api->abortStream();
 				}
 				/**
 				 * @return true if a stream is open and false if not.
 				 */
 				bool isStreamOpen() const {
-					if (m_rtapi == nullptr) {
+					if (m_api == nullptr) {
 						return false;
 					}
-					return m_rtapi->isStreamOpen();
+					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_rtapi == nullptr) {
+					if (m_api == nullptr) {
 						return false;
 					}
-					return m_rtapi->isStreamRunning();
+					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_rtapi == nullptr) {
+					if (m_api == nullptr) {
 						return audio::Time();
 					}
-					return m_rtapi->getStreamTime();
+					return m_api->getStreamTime();
 				}
 				/**
 				 * @brief The stream latency refers to delay in audio input and/or output
@@ -294,10 +294,10 @@ namespace audio {
 				 * @return The internal stream latency in sample frames.
 				 */
 				long getStreamLatency() {
-					if (m_rtapi == nullptr) {
+					if (m_api == nullptr) {
 						return 0;
 					}
-					return m_rtapi->getStreamLatency();
+					return m_api->getStreamLatency();
 				}
 				/**
 				 * @brief On some systems, the sample rate used may be slightly different
@@ -306,10 +306,10 @@ namespace audio {
 				 * @return Returns actual sample rate in use by the stream.
 				 */
 				uint32_t getStreamSampleRate() {
-					if (m_rtapi == nullptr) {
+					if (m_api == nullptr) {
 						return 0;
 					}
-					return m_rtapi->getStreamSampleRate();
+					return m_api->getStreamSampleRate();
 				}
 				bool isMasterOf(audio::orchestra::Interface& _interface);
 			protected:
diff --git a/audio/orchestra/api/Alsa.cpp b/audio/orchestra/api/Alsa.cpp
index 75cb896..8eeb2ca 100644
--- a/audio/orchestra/api/Alsa.cpp
+++ b/audio/orchestra/api/Alsa.cpp
@@ -104,7 +104,8 @@ uint32_t audio::orchestra::api::Alsa::getDeviceCount() {
 			if (subdevice < 0) {
 				break;
 			}
-			nDevices++;
+			// input/output
+			nDevices+=2;
 		}
 nextcard:
 		snd_ctl_close(handle);
@@ -113,7 +114,7 @@ nextcard:
 	return nDevices;
 }
 
-bool audio::orchestra::api::Alsa::getNamedDeviceInfoLocal(const std::string& _deviceName, audio::orchestra::DeviceInfo& _info, int32_t _cardId, int32_t _subdevice, int32_t _localDeviceId) {
+bool audio::orchestra::api::Alsa::getNamedDeviceInfoLocal(const std::string& _deviceName, audio::orchestra::DeviceInfo& _info, int32_t _cardId, int32_t _subdevice, int32_t _localDeviceId, bool _input) {
 	int32_t result;
 	snd_ctl_t *chandle;
 	int32_t openMode = SND_PCM_ASYNC;
@@ -124,7 +125,12 @@ bool audio::orchestra::api::Alsa::getNamedDeviceInfoLocal(const std::string& _de
 	snd_pcm_hw_params_t *params;
 	snd_pcm_hw_params_alloca(&params);
 	// First try for playback unless default _device (which has subdev -1)
-	stream = SND_PCM_STREAM_PLAYBACK;
+	_info.input = _input;
+	if (_input == true) {
+		stream = SND_PCM_STREAM_CAPTURE;
+	} else {
+		stream = SND_PCM_STREAM_PLAYBACK;
+	}
 	snd_pcm_info_set_stream(pcminfo, stream);
 	std::vector<std::string> listElement = etk::split(_deviceName, ',');
 	if (listElement.size() == 0) {
@@ -146,39 +152,14 @@ bool audio::orchestra::api::Alsa::getNamedDeviceInfoLocal(const std::string& _de
 		snd_pcm_info_set_subdevice(pcminfo, 0);
 		result = snd_ctl_pcm_info(chandle, pcminfo);
 		if (result < 0) {
-			// Device probably doesn't support playback.
-			goto captureProbe;
+			// Device probably doesn't support IO.
+			return false;
 		}
 	}
-	result = snd_pcm_open(&phandle, _deviceName.c_str(), stream, openMode | SND_PCM_NONBLOCK);
-	if (result < 0) {
-		ATA_ERROR("snd_pcm_open error for device (" << _deviceName << "), " << snd_strerror(result) << ". (seems to be already open)");
-		// TODO : Return audio::orchestra::error_warning;
-		goto captureProbe;
+	// ALSA doesn't provide default devices so we'll use the first available one.
+	if (_localDeviceId == 0) {
+		_info.isDefault = true;
 	}
-	// The device is open ... fill the parameter structure.
-	result = snd_pcm_hw_params_any(phandle, params);
-	if (result < 0) {
-		snd_pcm_close(phandle);
-		ATA_ERROR("snd_pcm_hw_params error for device (" << _deviceName << "), " << snd_strerror(result) << ".");
-		// TODO : Return audio::orchestra::error_warning;
-		goto captureProbe;
-	}
-	// Get output channel information.
-	uint32_t value;
-	result = snd_pcm_hw_params_get_channels_max(params, &value);
-	if (result < 0) {
-		snd_pcm_close(phandle);
-		ATA_ERROR("error getting device (" << _deviceName << ") output channels, " << snd_strerror(result) << ".");
-		// TODO : Return audio::orchestra::error_warning;
-		goto captureProbe;
-	}
-	ATA_ERROR("Output channel = " << value);
-	_info.outputChannels = value;
-	snd_pcm_close(phandle);
-
-captureProbe:
-	stream = SND_PCM_STREAM_CAPTURE;
 	snd_pcm_info_set_stream(pcminfo, stream);
 	// Now try for capture unless default device (with subdev = -1)
 	if (_subdevice != -1) {
@@ -186,82 +167,32 @@ captureProbe:
 		snd_ctl_close(chandle);
 		if (result < 0) {
 			// Device probably doesn't support capture.
-			if (_info.outputChannels == 0) {
-				return true;
-			}
-			goto probeParameters;
+			return false;
 		}
 	}
+	// open device:
 	result = snd_pcm_open(&phandle, _deviceName.c_str(), stream, openMode | SND_PCM_NONBLOCK);
 	if (result < 0) {
 		ATA_ERROR("snd_pcm_open error for device (" << _deviceName << "), " << snd_strerror(result) << ". (seems to be already open)");
-		// TODO : Return audio::orchestra::error_warning;
-		if (_info.outputChannels == 0) {
-			return true;
-		}
-		goto probeParameters;
+		return true;
 	}
 	// The device is open ... fill the parameter structure.
 	result = snd_pcm_hw_params_any(phandle, params);
 	if (result < 0) {
 		snd_pcm_close(phandle);
 		ATA_ERROR("snd_pcm_hw_params error for device (" << _deviceName << "), " << snd_strerror(result) << ".");
-		// TODO : Return audio::orchestra::error_warning;
-		if (_info.outputChannels == 0) {
-			return true;
-		}
-		goto probeParameters;
+		return false;
 	}
+	unsigned int value;
 	result = snd_pcm_hw_params_get_channels_max(params, &value);
 	if (result < 0) {
 		snd_pcm_close(phandle);
 		ATA_ERROR("error getting device (" << _deviceName << ") input channels, " << snd_strerror(result) << ".");
-		// TODO : Return audio::orchestra::error_warning;
-		if (_info.outputChannels == 0) {
-			return true;
-		}
-		goto probeParameters;
+		return false;
 	}
 	ATA_ERROR("Input channel = " << value);
-	_info.inputChannels = value;
-	snd_pcm_close(phandle);
-	// ALSA does not support duplex, but synchronization between I/Os
-	_info.duplexChannels = 0;
-	// ALSA doesn't provide default devices so we'll use the first available one.
-	if (    _localDeviceId == 0
-	     && _info.outputChannels > 0) {
-		_info.isDefaultOutput = true;
-	}
-	if (    _localDeviceId == 0
-	     && _info.inputChannels > 0) {
-		_info.isDefaultInput = true;
-	}
-
-probeParameters:
-	// At this point, we just need to figure out the supported data
-	// formats and sample rates. We'll proceed by opening the device in
-	// the direction with the maximum number of channels, or playback if
-	// they are equal. This might limit our sample rate options, but so
-	// be it.
-	if (_info.outputChannels >= _info.inputChannels) {
-		stream = SND_PCM_STREAM_PLAYBACK;
-	} else {
-		stream = SND_PCM_STREAM_CAPTURE;
-	}
-	snd_pcm_info_set_stream(pcminfo, stream);
-	result = snd_pcm_open(&phandle, _deviceName.c_str(), stream, openMode | SND_PCM_NONBLOCK);
-	if (result < 0) {
-		ATA_ERROR("snd_pcm_open error for device (" << _deviceName << "), " << snd_strerror(result) << ".");
-		// TODO : Return audio::orchestra::error_warning;
-		return false;
-	}
-	// The device is open ... fill the parameter structure.
-	result = snd_pcm_hw_params_any(phandle, params);
-	if (result < 0) {
-		snd_pcm_close(phandle);
-		ATA_ERROR("snd_pcm_hw_params error for device (" << _deviceName << "), " << snd_strerror(result) << ".");
-		// TODO : Return audio::orchestra::error_warning;
-		return false;
+	for (int32_t iii=0; iii<value; ++iii) {
+		_info.channels.push_back(audio::channel_unknow);
 	}
 	// Test our discrete set of sample rate values.
 	_info.sampleRates.clear();
@@ -275,7 +206,6 @@ probeParameters:
 	if (_info.sampleRates.size() == 0) {
 		snd_pcm_close(phandle);
 		ATA_ERROR("no supported sample rates found for device (" << _deviceName << ").");
-		// TODO : Return audio::orchestra::error_warning;
 		return false;
 	}
 	// Probe the supported data formats ... we don't care about endian-ness just yet
@@ -325,24 +255,16 @@ probeParameters:
 	}
 	// That's all ... close the device and return
 	snd_pcm_close(phandle);
-	_info.probed = true;
 	return true;
 }
 
 audio::orchestra::DeviceInfo audio::orchestra::api::Alsa::getDeviceInfo(uint32_t _device) {
 	audio::orchestra::DeviceInfo info;
-	/*
-	ATA_WARNING("plop");
-	getDeviceInfo("hw:0,0,0", info);
-	info.display();
-	getDeviceInfo("hw:0,0,1", info);
-	info.display();
-	*/
-	info.probed = false;
 	unsigned nDevices = 0;
 	int32_t result = -1;
 	int32_t subdevice = -1;
 	int32_t card = -1;
+	bool isInput = false;
 	char name[64];
 	snd_ctl_t *chandle;
 	// Count cards and devices
@@ -366,10 +288,18 @@ audio::orchestra::DeviceInfo audio::orchestra::api::Alsa::getDeviceInfo(uint32_t
 				break;
 			}
 			if (nDevices == _device) {
+				// for input
 				sprintf(name, "hw:%d,%d", card, subdevice);
+				isInput = true;
 				goto foundDevice;
 			}
-			nDevices++;
+			if (nDevices+1 == _device) {
+				// for output
+				sprintf(name, "hw:%d,%d", card, subdevice);
+				isInput = false;
+				goto foundDevice;
+			}
+			nDevices+=2;
 		}
 	nextcard:
 		snd_ctl_close(chandle);
@@ -398,9 +328,9 @@ foundDevice:
 			// TODO : return audio::orchestra::error_warning;
 			return info;
 		}
-		return m_devices[ _device ];
+		return m_devices[_device];
 	}
-	bool ret = audio::orchestra::api::Alsa::getNamedDeviceInfoLocal(name, info, card, subdevice, _device);
+	bool ret = audio::orchestra::api::Alsa::getNamedDeviceInfoLocal(name, info, card, subdevice, _device, isInput);
 	if (ret == false) {
 		// TODO : ...
 		return info;
@@ -418,13 +348,13 @@ void audio::orchestra::api::Alsa::saveDeviceInfo() {
 }
 
 bool audio::orchestra::api::Alsa::probeDeviceOpen(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) {
+                                                  audio::orchestra::mode _mode,
+                                                  uint32_t _channels,
+                                                  uint32_t _firstChannel,
+                                                  uint32_t _sampleRate,
+                                                  audio::format _format,
+                                                  uint32_t *_bufferSize,
+                                                  const audio::orchestra::StreamOptions& _options) {
 	// I'm not using the "plug" interface ... too much inconsistent behavior.
 	unsigned nDevices = 0;
 	int32_t result, subdevice, card;
diff --git a/audio/orchestra/api/Alsa.h b/audio/orchestra/api/Alsa.h
index a1380c1..3024ff9 100644
--- a/audio/orchestra/api/Alsa.h
+++ b/audio/orchestra/api/Alsa.h
@@ -27,7 +27,8 @@ namespace audio {
 					                             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 fined
+					                             int32_t _localDeviceId=-1,// local ID of device find
+					                             bool _input=false);
 				public:
 					bool getNamedDeviceInfo(const std::string& _deviceName, audio::orchestra::DeviceInfo& _info) {
 						return getNamedDeviceInfoLocal(_deviceName, _info);
@@ -48,7 +49,6 @@ namespace audio {
 					void callbackEventOneCycleMMAPWrite();
 				private:
 					static void alsaCallbackEvent(void* _userData);
-					static void alsaCallbackEventMMap(void* _userData);
 				private:
 					std11::shared_ptr<AlsaPrivate> m_private;
 					std::vector<audio::orchestra::DeviceInfo> m_devices;
diff --git a/audio/orchestra/api/Ds.cpp b/audio/orchestra/api/Ds.cpp
index 0617e5f..c668e0c 100644
--- a/audio/orchestra/api/Ds.cpp
+++ b/audio/orchestra/api/Ds.cpp
@@ -9,6 +9,7 @@
 #if defined(ORCHESTRA_BUILD_DS)
 #include <audio/orchestra/Interface.h>
 #include <audio/orchestra/debug.h>
+#include <etk/thread/tools.h>
 
 #undef __class__
 #define __class__ "api::Ds"
@@ -58,15 +59,13 @@ static inline DWORD dsPointerBetween(DWORD _pointer, DWORD _laterPointer, DWORD
 
 class DsDevice {
 	public:
-		LPGUID id[2];
-		bool validId[2];
-		bool found;
+		LPGUID id;
+		bool input;
 		std::string name;
-		
 		DsDevice() :
-		  found(false) {
-			validId[0] = false;
-			validId[1] = false;
+		  id(0),
+		  input(false) {
+			
 		}
 };
 
@@ -105,12 +104,6 @@ namespace audio {
 	}
 }
 
-// Declarations for utility functions, callbacks, and structures
-// specific to the DirectSound implementation.
-static BOOL CALLBACK deviceQueryCallback(LPGUID _lpguid,
-                                         LPCTSTR _description,
-                                         LPCTSTR _module,
-                                         LPVOID _lpContext);
 
 static const char* getErrorString(int32_t _code);
 
@@ -139,220 +132,248 @@ audio::orchestra::api::Ds::~Ds() {
 	}
 }
 
-// The DirectSound default output is always the first device.
-uint32_t audio::orchestra::api::Ds::getDefaultOutputDevice() {
-	return 0;
+
+#include "tchar.h"
+static std::string convertTChar(LPCTSTR _name) {
+#if defined(UNICODE) || defined(_UNICODE)
+	int32_t length = WideCharToMultiByte(CP_UTF8, 0, _name, -1, nullptr, 0, nullptr, nullptr);
+	std::string s(length-1, '\0');
+	WideCharToMultiByte(CP_UTF8, 0, _name, -1, &s[0], length, nullptr, nullptr);
+#else
+	std::string s(_name);
+#endif
+	return s;
 }
 
-// The DirectSound default input is always the first input device,
-// which is the first capture device enumerated.
-uint32_t audio::orchestra::api::Ds::getDefaultInputDevice() {
-	return 0;
+static BOOL CALLBACK deviceQueryCallback(LPGUID _lpguid,
+                                         LPCTSTR _description,
+                                         LPCTSTR _module,
+                                         LPVOID _lpContext) {
+	struct DsProbeData& probeInfo = *(struct DsProbeData*) _lpContext;
+	std::vector<DsDevice>& dsDevices = *probeInfo.dsDevices;
+	HRESULT hr;
+	bool validDevice = false;
+	if (probeInfo.isInput == true) {
+		DSCCAPS caps;
+		LPDIRECTSOUNDCAPTURE object;
+		hr = DirectSoundCaptureCreate(_lpguid, &object,	 nullptr);
+		if (hr != DS_OK) {
+			return TRUE;
+		}
+		caps.dwSize = sizeof(caps);
+		hr = object->GetCaps(&caps);
+		if (hr == DS_OK) {
+			if (caps.dwChannels > 0 && caps.dwFormats > 0) {
+				validDevice = true;
+			}
+		}
+		object->Release();
+	} else {
+		DSCAPS caps;
+		LPDIRECTSOUND object;
+		hr = DirectSoundCreate(_lpguid, &object,	 nullptr);
+		if (hr != DS_OK) {
+			return TRUE;
+		}
+		caps.dwSize = sizeof(caps);
+		hr = object->GetCaps(&caps);
+		if (hr == DS_OK) {
+			if (    caps.dwFlags & DSCAPS_PRIMARYMONO
+			     || caps.dwFlags & DSCAPS_PRIMARYSTEREO) {
+				validDevice = true;
+			}
+		}
+		object->Release();
+	}
+	if (validDevice == false) {
+		return TRUE;
+	}
+	// If good device, then save its name and guid.
+	std::string name = convertTChar(_description);
+	//if (name == "Primary Sound Driver" || name == "Primary Sound Capture Driver")
+	if (_lpguid == nullptr) {
+		name = "Default Device";
+	}
+	DsDevice device;
+	device.name = name;
+	device.input = probeInfo.isInput;
+	device.id = _lpguid;
+	dsDevices.push_back(device);
+	return TRUE;
 }
 
 uint32_t audio::orchestra::api::Ds::getDeviceCount() {
-	// Set query flag for previously found devices to false, so that we
-	// can check for any devices that have disappeared.
-	for (size_t iii=0; iii<m_private->dsDevices.size(); ++iii) {
-		m_private->dsDevices[iii].found = false;
-	}
 	// Query DirectSound devices.
 	struct DsProbeData probeInfo;
 	probeInfo.isInput = false;
 	probeInfo.dsDevices = &m_private->dsDevices;
 	HRESULT result = DirectSoundEnumerate((LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo);
 	if (FAILED(result)) {
-		ATA_ERROR("error (" << getErrorString(result) << ") enumerating output devices!");
+		ATA_ERROR(getErrorString(result) << ": enumerating output devices!");
 		return 0;
 	}
 	// Query DirectSoundCapture devices.
 	probeInfo.isInput = true;
 	result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo);
 	if (FAILED(result)) {
-		ATA_ERROR("error (" << getErrorString(result) << ") enumerating input devices!");
+		ATA_ERROR(getErrorString(result) << ": enumerating input devices!");
 		return 0;
 	}
-	// Clean out any devices that may have disappeared.
-	std::vector< int32_t > indices;
-	for (uint32_t i=0; i<m_private->dsDevices.size(); i++) {
-		if (m_private->dsDevices[i].found == false) {
-			indices.push_back(i);
-		}
-	}
-	uint32_t nErased = 0;
-	for (uint32_t i=0; i<indices.size(); i++) {
-		m_private->dsDevices.erase(m_private->dsDevices.begin()-nErased++);
-	}
 	return m_private->dsDevices.size();
 }
 
 audio::orchestra::DeviceInfo audio::orchestra::api::Ds::getDeviceInfo(uint32_t _device) {
 	audio::orchestra::DeviceInfo info;
-	info.probed = false;
-	if (m_private->dsDevices.size() == 0) {
-		// Force a query of all devices
-		getDeviceCount();
-		if (m_private->dsDevices.size() == 0) {
-			ATA_ERROR("no devices found!");
-			return info;
-		}
-	}
 	if (_device >= m_private->dsDevices.size()) {
 		ATA_ERROR("device ID is invalid!");
 		return info;
 	}
 	HRESULT result;
-	if (m_private->dsDevices[ _device ].validId[0] == false) {
-		goto probeInput;
-	}
-	LPDIRECTSOUND output;
-	DSCAPS outCaps;
-	result = DirectSoundCreate(m_private->dsDevices[ _device ].id[0], &output, nullptr);
-	if (FAILED(result)) {
-		ATA_ERROR("error (" << getErrorString(result) << ") opening output device (" << m_private->dsDevices[ _device ].name << ")!");
-		goto probeInput;
-	}
-	outCaps.dwSize = sizeof(outCaps);
-	result = output->GetCaps(&outCaps);
-	if (FAILED(result)) {
-		output->Release();
-		ATA_ERROR("error (" << getErrorString(result) << ") getting capabilities!");
-		goto probeInput;
-	}
-	// Get output channel information.
-	info.outputChannels = (outCaps.dwFlags & DSCAPS_PRIMARYSTEREO) ? 2 : 1;
-	// Get sample rate information.
-	info.sampleRates.clear();
-	for (auto &it : audio::orchestra::genericSampleRate()) {
-		if (    it >= outCaps.dwMinSecondarySampleRate
-		     && it <= outCaps.dwMaxSecondarySampleRate) {
-			info.sampleRates.push_back(it);
+	if (m_private->dsDevices[_device].input == false) {
+		LPDIRECTSOUND output;
+		DSCAPS outCaps;
+		result = DirectSoundCreate(m_private->dsDevices[_device].id, &output, nullptr);
+		if (FAILED(result)) {
+			ATA_ERROR(getErrorString(result) << ": opening output device (" << m_private->dsDevices[_device].name << ")!");
+			return info;
 		}
-	}
-	// Get format information.
-	if (outCaps.dwFlags & DSCAPS_PRIMARY16BIT) {
-		info.nativeFormats.push_back(audio::format_int16);
-	}
-	if (outCaps.dwFlags & DSCAPS_PRIMARY8BIT) {
-		info.nativeFormats.push_back(audio::format_int8);
-	}
-	output->Release();
-	if (getDefaultOutputDevice() == _device) {
-		info.isDefaultOutput = true;
-	}
-	if (m_private->dsDevices[ _device ].validId[1] == false) {
-		info.name = m_private->dsDevices[ _device ].name;
-		info.probed = true;
-		return info;
-	}
-probeInput:
-	LPDIRECTSOUNDCAPTURE input;
-	result = DirectSoundCaptureCreate(m_private->dsDevices[ _device ].id[1], &input, nullptr);
-	if (FAILED(result)) {
-		ATA_ERROR("error (" << getErrorString(result) << ") opening input device (" << m_private->dsDevices[ _device ].name << ")!");
-		return info;
-	}
-	DSCCAPS inCaps;
-	inCaps.dwSize = sizeof(inCaps);
-	result = input->GetCaps(&inCaps);
-	if (FAILED(result)) {
-		input->Release();
-		ATA_ERROR("error (" << getErrorString(result) << ") getting object capabilities (" << m_private->dsDevices[ _device ].name << ")!");
-		return info;
-	}
-	// Get input channel information.
-	info.inputChannels = inCaps.dwChannels;
-	// Get sample rate and format information.
-	std::vector<uint32_t> rates;
-	if (inCaps.dwChannels >= 2) {
-		if (    (inCaps.dwFormats & WAVE_FORMAT_1S16)
-		     || (inCaps.dwFormats & WAVE_FORMAT_2S16)
-		     || (inCaps.dwFormats & WAVE_FORMAT_4S16)
-		     || (inCaps.dwFormats & WAVE_FORMAT_96S16) ) {
-			info.nativeFormats.push_back(audio::format_int16);
+		outCaps.dwSize = sizeof(outCaps);
+		result = output->GetCaps(&outCaps);
+		if (FAILED(result)) {
+			output->Release();
+			ATA_ERROR(getErrorString(result) << ": getting capabilities!");
+			return info;
 		}
-		if (    (inCaps.dwFormats & WAVE_FORMAT_1S08)
-		     || (inCaps.dwFormats & WAVE_FORMAT_2S08)
-		     || (inCaps.dwFormats & WAVE_FORMAT_4S08)
-		     || (inCaps.dwFormats & WAVE_FORMAT_96S08) ) {
-			info.nativeFormats.push_back(audio::format_int8);
+		// Get output channel information.
+		if (outCaps.dwFlags & DSCAPS_PRIMARYSTEREO) {
+			info.channels.push_back(audio::channel_unknow);
+			info.channels.push_back(audio::channel_unknow);
+		} else {
+			info.channels.push_back(audio::channel_unknow);
 		}
-		if (    (inCaps.dwFormats & WAVE_FORMAT_1S16)
-		     || (inCaps.dwFormats & WAVE_FORMAT_1S08) ){
-			rates.push_back(11025);
-		}
-		if (    (inCaps.dwFormats & WAVE_FORMAT_2S16)
-		     || (inCaps.dwFormats & WAVE_FORMAT_2S08) ){
-			rates.push_back(22050);
-		}
-		if (    (inCaps.dwFormats & WAVE_FORMAT_4S16)
-		     || (inCaps.dwFormats & WAVE_FORMAT_4S08) ){
-			rates.push_back(44100);
-		}
-		if (    (inCaps.dwFormats & WAVE_FORMAT_96S16)
-		     || (inCaps.dwFormats & WAVE_FORMAT_96S08) ){
-			rates.push_back(96000);
-		}
-	} else if (inCaps.dwChannels == 1) {
-		if (    (inCaps.dwFormats & WAVE_FORMAT_1M16)
-		     || (inCaps.dwFormats & WAVE_FORMAT_2M16)
-		     || (inCaps.dwFormats & WAVE_FORMAT_4M16)
-		     || (inCaps.dwFormats & WAVE_FORMAT_96M16) ) {
-			info.nativeFormats.push_back(audio::format_int16);
-		}
-		if (    (inCaps.dwFormats & WAVE_FORMAT_1M08)
-		     || (inCaps.dwFormats & WAVE_FORMAT_2M08)
-		     || (inCaps.dwFormats & WAVE_FORMAT_4M08)
-		     || (inCaps.dwFormats & WAVE_FORMAT_96M08) ) {
-			info.nativeFormats.push_back(audio::format_int8);
-		}
-		if (    (inCaps.dwFormats & WAVE_FORMAT_1M16)
-		     || (inCaps.dwFormats & WAVE_FORMAT_1M08) ){
-			rates.push_back(11025);
-		}
-		if (    (inCaps.dwFormats & WAVE_FORMAT_2M16)
-		     || (inCaps.dwFormats & WAVE_FORMAT_2M08) ){
-			rates.push_back(22050);
-		}
-		if (    (inCaps.dwFormats & WAVE_FORMAT_4M16)
-		     || (inCaps.dwFormats & WAVE_FORMAT_4M08) ){
-			rates.push_back(44100);
-		}
-		if (    (inCaps.dwFormats & WAVE_FORMAT_96M16)
-		     || (inCaps.dwFormats & WAVE_FORMAT_96M08) ){
-			rates.push_back(96000);
-		}
-	} else {
-		// technically, this would be an error
-		info.inputChannels = 0;
-	}
-	input->Release();
-	if (info.inputChannels == 0) {
-		return info;
-	}
-	// Copy the supported rates to the info structure but avoid duplication.
-	bool found;
-	for (uint32_t i=0; i<rates.size(); i++) {
-		found = false;
-		for (uint32_t j=0; j<info.sampleRates.size(); j++) {
-			if (rates[i] == info.sampleRates[j]) {
-				found = true;
-				break;
+		// Get sample rate information.
+		for (auto &it : audio::orchestra::genericSampleRate()) {
+			if (    it >= outCaps.dwMinSecondarySampleRate
+			     && it <= outCaps.dwMaxSecondarySampleRate) {
+				info.sampleRates.push_back(it);
 			}
 		}
-		if (found == false) info.sampleRates.push_back(rates[i]);
+		// Get format information.
+		if (outCaps.dwFlags & DSCAPS_PRIMARY16BIT) {
+			info.nativeFormats.push_back(audio::format_int16);
+		}
+		if (outCaps.dwFlags & DSCAPS_PRIMARY8BIT) {
+			info.nativeFormats.push_back(audio::format_int8);
+		}
+		output->Release();
+		info.name = m_private->dsDevices[_device].name;
+		return info;
+	} else {
+		LPDIRECTSOUNDCAPTURE input;
+		result = DirectSoundCaptureCreate(m_private->dsDevices[_device].id, &input, nullptr);
+		if (FAILED(result)) {
+			ATA_ERROR(getErrorString(result) << ": opening input device (" << m_private->dsDevices[_device].name << ")!");
+			return info;
+		}
+		DSCCAPS inCaps;
+		inCaps.dwSize = sizeof(inCaps);
+		result = input->GetCaps(&inCaps);
+		if (FAILED(result)) {
+			input->Release();
+			ATA_ERROR(getErrorString(result) << ": getting object capabilities (" << m_private->dsDevices[_device].name << ")!");
+			return info;
+		}
+		// Get input channel information.
+		for (int32_t iii=0; iii<inCaps.dwChannels; ++iii) {
+			info.channels.push_back(audio::channel_unknow);
+		}
+		// Get sample rate and format information.
+		std::vector<uint32_t> rates;
+		if (inCaps.dwChannels >= 2) {
+			if (    (inCaps.dwFormats & WAVE_FORMAT_1S16)
+			     || (inCaps.dwFormats & WAVE_FORMAT_2S16)
+			     || (inCaps.dwFormats & WAVE_FORMAT_4S16)
+			     || (inCaps.dwFormats & WAVE_FORMAT_96S16) ) {
+				info.nativeFormats.push_back(audio::format_int16);
+			}
+			if (    (inCaps.dwFormats & WAVE_FORMAT_1S08)
+			     || (inCaps.dwFormats & WAVE_FORMAT_2S08)
+			     || (inCaps.dwFormats & WAVE_FORMAT_4S08)
+			     || (inCaps.dwFormats & WAVE_FORMAT_96S08) ) {
+				info.nativeFormats.push_back(audio::format_int8);
+			}
+			if (    (inCaps.dwFormats & WAVE_FORMAT_1S16)
+			     || (inCaps.dwFormats & WAVE_FORMAT_1S08) ){
+				rates.push_back(11025);
+			}
+			if (    (inCaps.dwFormats & WAVE_FORMAT_2S16)
+			     || (inCaps.dwFormats & WAVE_FORMAT_2S08) ){
+				rates.push_back(22050);
+			}
+			if (    (inCaps.dwFormats & WAVE_FORMAT_4S16)
+			     || (inCaps.dwFormats & WAVE_FORMAT_4S08) ){
+				rates.push_back(44100);
+			}
+			if (    (inCaps.dwFormats & WAVE_FORMAT_96S16)
+			     || (inCaps.dwFormats & WAVE_FORMAT_96S08) ){
+				rates.push_back(96000);
+			}
+		} else if (inCaps.dwChannels == 1) {
+			if (    (inCaps.dwFormats & WAVE_FORMAT_1M16)
+			     || (inCaps.dwFormats & WAVE_FORMAT_2M16)
+			     || (inCaps.dwFormats & WAVE_FORMAT_4M16)
+			     || (inCaps.dwFormats & WAVE_FORMAT_96M16) ) {
+				info.nativeFormats.push_back(audio::format_int16);
+			}
+			if (    (inCaps.dwFormats & WAVE_FORMAT_1M08)
+			     || (inCaps.dwFormats & WAVE_FORMAT_2M08)
+			     || (inCaps.dwFormats & WAVE_FORMAT_4M08)
+			     || (inCaps.dwFormats & WAVE_FORMAT_96M08) ) {
+				info.nativeFormats.push_back(audio::format_int8);
+			}
+			if (    (inCaps.dwFormats & WAVE_FORMAT_1M16)
+			     || (inCaps.dwFormats & WAVE_FORMAT_1M08) ){
+				rates.push_back(11025);
+			}
+			if (    (inCaps.dwFormats & WAVE_FORMAT_2M16)
+			     || (inCaps.dwFormats & WAVE_FORMAT_2M08) ){
+				rates.push_back(22050);
+			}
+			if (    (inCaps.dwFormats & WAVE_FORMAT_4M16)
+			     || (inCaps.dwFormats & WAVE_FORMAT_4M08) ){
+				rates.push_back(44100);
+			}
+			if (    (inCaps.dwFormats & WAVE_FORMAT_96M16)
+			     || (inCaps.dwFormats & WAVE_FORMAT_96M08) ){
+				rates.push_back(96000);
+			}
+		} else {
+			// technically, this would be an error
+			info.channels.clear();
+		}
+		input->Release();
+		if (info.channels.size() == 0) {
+			return info;
+		}
+		// Copy the supported rates to the info structure but avoid duplication.
+		bool found;
+		for (uint32_t i=0; i<rates.size(); i++) {
+			found = false;
+			for (uint32_t j=0; j<info.sampleRates.size(); j++) {
+				if (rates[i] == info.sampleRates[j]) {
+					found = true;
+					break;
+				}
+			}
+			if (found == false) {
+				info.sampleRates.push_back(rates[i]);
+			}
+		}
+		std::sort(info.sampleRates.begin(), info.sampleRates.end());
+		// Copy name and return.
+		info.name = m_private->dsDevices[_device].name;
+		return info;
 	}
-	std::sort(info.sampleRates.begin(), info.sampleRates.end());
-	// If device opens for both playback and capture, we determine the channels.
-	if (info.outputChannels > 0 && info.inputChannels > 0) {
-		info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
-	}
-	if (_device == 0) {
-		info.isDefaultInput = true;
-	}
-	// Copy name and return.
-	info.name = m_private->dsDevices[ _device ].name;
-	info.probed = true;
 	return info;
 }
 
@@ -379,17 +400,6 @@ bool audio::orchestra::api::Ds::probeDeviceOpen(uint32_t _device,
 		ATA_ERROR("device ID is invalid!");
 		return false;
 	}
-	if (_mode == audio::orchestra::mode_output) {
-		if (m_private->dsDevices[ _device ].validId[0] == false) {
-			ATA_ERROR("device (" << _device << ") does not support output!");
-			return false;
-		}
-	} else { // _mode == audio::orchestra::mode_input
-		if (m_private->dsDevices[ _device ].validId[1] == false) {
-			ATA_ERROR("device (" << _device << ") does not support input!");
-			return false;
-		}
-	}
 	// According to a note in PortAudio, using GetDesktopWindow()
 	// instead of GetForegroundWindow() is supposed to avoid problems
 	// that occur when the application's window is not the foreground
@@ -433,9 +443,9 @@ bool audio::orchestra::api::Ds::probeDeviceOpen(uint32_t _device,
 	HRESULT result;
 	if (_mode == audio::orchestra::mode_output) {
 		LPDIRECTSOUND output;
-		result = DirectSoundCreate(m_private->dsDevices[ _device ].id[0], &output, nullptr);
+		result = DirectSoundCreate(m_private->dsDevices[_device].id, &output, nullptr);
 		if (FAILED(result)) {
-			ATA_ERROR("error (" << getErrorString(result) << ") opening output device (" << m_private->dsDevices[ _device ].name << ")!");
+			ATA_ERROR(getErrorString(result) << ": opening output device (" << m_private->dsDevices[_device].name << ")!");
 			return false;
 		}
 		DSCAPS outCaps;
@@ -443,12 +453,12 @@ bool audio::orchestra::api::Ds::probeDeviceOpen(uint32_t _device,
 		result = output->GetCaps(&outCaps);
 		if (FAILED(result)) {
 			output->Release();
-			ATA_ERROR("error (" << getErrorString(result) << ") getting capabilities (" << m_private->dsDevices[ _device ].name << ")!");
+			ATA_ERROR(getErrorString(result) << ": getting capabilities (" << m_private->dsDevices[_device].name << ")!");
 			return false;
 		}
 		// Check channel information.
 		if (_channels + _firstChannel == 2 && !(outCaps.dwFlags & DSCAPS_PRIMARYSTEREO)) {
-			ATA_ERROR("the output device (" << m_private->dsDevices[ _device ].name << ") does not support stereo playback.");
+			ATA_ERROR("the output device (" << m_private->dsDevices[_device].name << ") does not support stereo playback.");
 			return false;
 		}
 		// Check format information.	Use 16-bit format unless not
@@ -477,7 +487,7 @@ bool audio::orchestra::api::Ds::probeDeviceOpen(uint32_t _device,
 		result = output->SetCooperativeLevel(hWnd, DSSCL_PRIORITY);
 		if (FAILED(result)) {
 			output->Release();
-			ATA_ERROR("error (" << getErrorString(result) << ") setting cooperative level (" << m_private->dsDevices[ _device ].name << ")!");
+			ATA_ERROR(getErrorString(result) << ": setting cooperative level (" << m_private->dsDevices[_device].name << ")!");
 			return false;
 		}
 		// Even though we will write to the secondary buffer, we need to
@@ -493,14 +503,14 @@ bool audio::orchestra::api::Ds::probeDeviceOpen(uint32_t _device,
 		result = output->CreateSoundBuffer(&bufferDescription, &buffer, nullptr);
 		if (FAILED(result)) {
 			output->Release();
-			ATA_ERROR("error (" << getErrorString(result) << ") accessing primary buffer (" << m_private->dsDevices[ _device ].name << ")!");
+			ATA_ERROR(getErrorString(result) << ": accessing primary buffer (" << m_private->dsDevices[_device].name << ")!");
 			return false;
 		}
 		// Set the primary DS buffer sound format.
 		result = buffer->SetFormat(&waveFormat);
 		if (FAILED(result)) {
 			output->Release();
-			ATA_ERROR("error (" << getErrorString(result) << ") setting primary buffer format (" << m_private->dsDevices[ _device ].name << ")!");
+			ATA_ERROR(getErrorString(result) << ": setting primary buffer format (" << m_private->dsDevices[_device].name << ")!");
 			return false;
 		}
 		// Setup the secondary DS buffer description.
@@ -523,7 +533,7 @@ bool audio::orchestra::api::Ds::probeDeviceOpen(uint32_t _device,
 			result = output->CreateSoundBuffer(&bufferDescription, &buffer, nullptr);
 			if (FAILED(result)) {
 				output->Release();
-				ATA_ERROR("error (" << getErrorString(result) << ") creating secondary buffer (" << m_private->dsDevices[ _device ].name << ")!");
+				ATA_ERROR(getErrorString(result) << ": creating secondary buffer (" << m_private->dsDevices[_device].name << ")!");
 				return false;
 			}
 		}
@@ -534,7 +544,7 @@ bool audio::orchestra::api::Ds::probeDeviceOpen(uint32_t _device,
 		if (FAILED(result)) {
 			output->Release();
 			buffer->Release();
-			ATA_ERROR("error (" << getErrorString(result) << ") getting buffer settings (" << m_private->dsDevices[ _device ].name << ")!");
+			ATA_ERROR(getErrorString(result) << ": getting buffer settings (" << m_private->dsDevices[_device].name << ")!");
 			return false;
 		}
 		dsBufferSize = dsbcaps.dwBufferBytes;
@@ -545,7 +555,7 @@ bool audio::orchestra::api::Ds::probeDeviceOpen(uint32_t _device,
 		if (FAILED(result)) {
 			output->Release();
 			buffer->Release();
-			ATA_ERROR("error (" << getErrorString(result) << ") locking buffer (" << m_private->dsDevices[ _device ].name << ")!");
+			ATA_ERROR(getErrorString(result) << ": locking buffer (" << m_private->dsDevices[_device].name << ")!");
 			return false;
 		}
 		// Zero the DS buffer
@@ -555,7 +565,7 @@ bool audio::orchestra::api::Ds::probeDeviceOpen(uint32_t _device,
 		if (FAILED(result)) {
 			output->Release();
 			buffer->Release();
-			ATA_ERROR("error (" << getErrorString(result) << ") unlocking buffer (" << m_private->dsDevices[ _device ].name << ")!");
+			ATA_ERROR(getErrorString(result) << ": unlocking buffer (" << m_private->dsDevices[_device].name << ")!");
 			return false;
 		}
 		ohandle = (void *) output;
@@ -563,9 +573,9 @@ bool audio::orchestra::api::Ds::probeDeviceOpen(uint32_t _device,
 	}
 	if (_mode == audio::orchestra::mode_input) {
 		LPDIRECTSOUNDCAPTURE input;
-		result = DirectSoundCaptureCreate(m_private->dsDevices[ _device ].id[1], &input, nullptr);
+		result = DirectSoundCaptureCreate(m_private->dsDevices[_device].id, &input, nullptr);
 		if (FAILED(result)) {
-			ATA_ERROR("error (" << getErrorString(result) << ") opening input device (" << m_private->dsDevices[ _device ].name << ")!");
+			ATA_ERROR(getErrorString(result) << ": opening input device (" << m_private->dsDevices[_device].name << ")!");
 			return false;
 		}
 		DSCCAPS inCaps;
@@ -573,7 +583,7 @@ bool audio::orchestra::api::Ds::probeDeviceOpen(uint32_t _device,
 		result = input->GetCaps(&inCaps);
 		if (FAILED(result)) {
 			input->Release();
-			ATA_ERROR("error (" << getErrorString(result) << ") getting input capabilities (" << m_private->dsDevices[ _device ].name << ")!");
+			ATA_ERROR(getErrorString(result) << ": getting input capabilities (" << m_private->dsDevices[_device].name << ")!");
 			return false;
 		}
 		// Check channel information.
@@ -626,7 +636,7 @@ bool audio::orchestra::api::Ds::probeDeviceOpen(uint32_t _device,
 		result = input->CreateCaptureBuffer(&bufferDescription, &buffer, nullptr);
 		if (FAILED(result)) {
 			input->Release();
-			ATA_ERROR("error (" << getErrorString(result) << ") creating input buffer (" << m_private->dsDevices[ _device ].name << ")!");
+			ATA_ERROR(getErrorString(result) << ": creating input buffer (" << m_private->dsDevices[_device].name << ")!");
 			return false;
 		}
 		// Get the buffer size ... might be different from what we specified.
@@ -636,7 +646,7 @@ bool audio::orchestra::api::Ds::probeDeviceOpen(uint32_t _device,
 		if (FAILED(result)) {
 			input->Release();
 			buffer->Release();
-			ATA_ERROR("error (" << getErrorString(result) << ") getting buffer settings (" << m_private->dsDevices[ _device ].name << ")!");
+			ATA_ERROR(getErrorString(result) << ": getting buffer settings (" << m_private->dsDevices[_device].name << ")!");
 			return false;
 		}
 		dsBufferSize = dscbcaps.dwBufferBytes;
@@ -651,7 +661,7 @@ bool audio::orchestra::api::Ds::probeDeviceOpen(uint32_t _device,
 		if (FAILED(result)) {
 			input->Release();
 			buffer->Release();
-			ATA_ERROR("error (" << getErrorString(result) << ") locking input buffer (" << m_private->dsDevices[ _device ].name << ")!");
+			ATA_ERROR(getErrorString(result) << ": locking input buffer (" << m_private->dsDevices[_device].name << ")!");
 			return false;
 		}
 		// Zero the buffer
@@ -661,7 +671,7 @@ bool audio::orchestra::api::Ds::probeDeviceOpen(uint32_t _device,
 		if (FAILED(result)) {
 			input->Release();
 			buffer->Release();
-			ATA_ERROR("error (" << getErrorString(result) << ") unlocking input buffer (" << m_private->dsDevices[ _device ].name << ")!");
+			ATA_ERROR(getErrorString(result) << ": unlocking input buffer (" << m_private->dsDevices[_device].name << ")!");
 			return false;
 		}
 		ohandle = (void *) input;
@@ -749,7 +759,7 @@ bool audio::orchestra::api::Ds::probeDeviceOpen(uint32_t _device,
 			goto error;
 		}
 		// Boost DS thread priority
-		SetThreadPriority((HANDLE)m_private->thread, THREAD_PRIORITY_HIGHEST);
+		etk::thread::setPriority(*m_private->thread, -6);
 	}
 	return true;
 error:
@@ -788,8 +798,10 @@ enum audio::orchestra::error audio::orchestra::api::Ds::closeStream() {
 	}
 	// Stop the callback thread.
 	m_private->threadRunning = false;
-	WaitForSingleObject((HANDLE) m_private->thread, INFINITE);
-	CloseHandle((HANDLE) m_private->thread);
+	if (m_private->thread != nullptr) {
+		m_private->thread->join();
+		m_private->thread = nullptr;
+	}
 	if (m_private->buffer[0]) { // the object pointer can be nullptr and valid
 		LPDIRECTSOUND object = (LPDIRECTSOUND) m_private->id[0];
 		LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) m_private->buffer[0];
@@ -846,7 +858,7 @@ enum audio::orchestra::error audio::orchestra::api::Ds::startStream() {
 		LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) m_private->buffer[0];
 		result = buffer->Play(0, 0, DSBPLAY_LOOPING);
 		if (FAILED(result)) {
-			ATA_ERROR("error (" << getErrorString(result) << ") starting output buffer!");
+			ATA_ERROR(getErrorString(result) << ": starting output buffer!");
 			goto unlock;
 		}
 	}
@@ -855,7 +867,7 @@ enum audio::orchestra::error audio::orchestra::api::Ds::startStream() {
 		LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) m_private->buffer[1];
 		result = buffer->Start(DSCBSTART_LOOPING);
 		if (FAILED(result)) {
-			ATA_ERROR("error (" << getErrorString(result) << ") starting input buffer!");
+			ATA_ERROR(getErrorString(result) << ": starting input buffer!");
 			goto unlock;
 		}
 	}
@@ -892,14 +904,14 @@ enum audio::orchestra::error audio::orchestra::api::Ds::stopStream() {
 		LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) m_private->buffer[0];
 		result = buffer->Stop();
 		if (FAILED(result)) {
-			ATA_ERROR("error (" << getErrorString(result) << ") stopping output buffer!");
+			ATA_ERROR(getErrorString(result) << ": stopping output buffer!");
 			goto unlock;
 		}
 		// Lock the buffer and clear it so that if we start to play again,
 		// we won't have old data playing.
 		result = buffer->Lock(0, m_private->dsBufferSize[0], &audioPtr, &dataLen, nullptr, nullptr, 0);
 		if (FAILED(result)) {
-			ATA_ERROR("error (" << getErrorString(result) << ") locking output buffer!");
+			ATA_ERROR(getErrorString(result) << ": locking output buffer!");
 			goto unlock;
 		}
 		// Zero the DS buffer
@@ -907,7 +919,7 @@ enum audio::orchestra::error audio::orchestra::api::Ds::stopStream() {
 		// Unlock the DS buffer
 		result = buffer->Unlock(audioPtr, dataLen, nullptr, 0);
 		if (FAILED(result)) {
-			ATA_ERROR("error (" << getErrorString(result) << ") unlocking output buffer!");
+			ATA_ERROR(getErrorString(result) << ": unlocking output buffer!");
 			goto unlock;
 		}
 		// If we start playing again, we must begin at beginning of buffer.
@@ -921,14 +933,14 @@ enum audio::orchestra::error audio::orchestra::api::Ds::stopStream() {
 		m_state = audio::orchestra::state_stopped;
 		result = buffer->Stop();
 		if (FAILED(result)) {
-			ATA_ERROR("error (" << getErrorString(result) << ") stopping input buffer!");
+			ATA_ERROR(getErrorString(result) << ": stopping input buffer!");
 			goto unlock;
 		}
 		// Lock the buffer and clear it so that if we start to play again,
 		// we won't have old data playing.
 		result = buffer->Lock(0, m_private->dsBufferSize[1], &audioPtr, &dataLen, nullptr, nullptr, 0);
 		if (FAILED(result)) {
-			ATA_ERROR("error (" << getErrorString(result) << ") locking input buffer!");
+			ATA_ERROR(getErrorString(result) << ": locking input buffer!");
 			goto unlock;
 		}
 		// Zero the DS buffer
@@ -936,7 +948,7 @@ enum audio::orchestra::error audio::orchestra::api::Ds::stopStream() {
 		// Unlock the DS buffer
 		result = buffer->Unlock(audioPtr, dataLen, nullptr, 0);
 		if (FAILED(result)) {
-			ATA_ERROR("error (" << getErrorString(result) << ") unlocking input buffer!");
+			ATA_ERROR(getErrorString(result) << ": unlocking input buffer!");
 			goto unlock;
 		}
 		// If we start recording again, we must begin at beginning of buffer.
@@ -963,6 +975,7 @@ enum audio::orchestra::error audio::orchestra::api::Ds::abortStream() {
 }
 
 void audio::orchestra::api::Ds::callbackEvent() {
+	etk::thread::setName("DS IO-" + m_name);
 	if (m_state == audio::orchestra::state_stopped || m_state == audio::orchestra::state_stopping) {
 		Sleep(50); // sleep 50 milliseconds
 		return;
@@ -985,23 +998,23 @@ void audio::orchestra::api::Ds::callbackEvent() {
 	// draining stream.
 	if (m_private->drainCounter == 0) {
 		audio::Time streamTime = getStreamTime();
-		audio::orchestra::status status = audio::orchestra::status_ok;
+		std::vector<audio::orchestra::status> status;
 		if (    m_mode != audio::orchestra::mode_input
 		     && m_private->xrun[0] == true) {
-			status = audio::orchestra::status_underflow;
+			status.push_back(audio::orchestra::status_underflow);
 			m_private->xrun[0] = false;
 		}
 		if (    m_mode != audio::orchestra::mode_output
 		     && m_private->xrun[1] == true) {
-			status = audio::orchestra::status_overflow;
+			status.push_back(audio::orchestra::status_overflow);
 			m_private->xrun[1] = false;
 		}
-		int32_t cbReturnValue = info->callback(&m_userBuffer[1][0],
-		                                       streamTime,
-		                                       &m_userBuffer[0][0],
-		                                       streamTime,
-		                                       m_bufferSize,
-		                                       status);
+		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;
@@ -1042,23 +1055,23 @@ void audio::orchestra::api::Ds::callbackEvent() {
 			DWORD startSafeWritePointer, startSafeReadPointer;
 			result = dsWriteBuffer->GetCurrentPosition(nullptr, &startSafeWritePointer);
 			if (FAILED(result)) {
-				ATA_ERROR("error (" << getErrorString(result) << ") getting current write position!");
+				ATA_ERROR(getErrorString(result) << ": getting current write position!");
 				return;
 			}
 			result = dsCaptureBuffer->GetCurrentPosition(nullptr, &startSafeReadPointer);
 			if (FAILED(result)) {
-				ATA_ERROR("error (" << getErrorString(result) << ") getting current read position!");
+				ATA_ERROR(getErrorString(result) << ": getting current read position!");
 				return;
 			}
 			while (true) {
 				result = dsWriteBuffer->GetCurrentPosition(nullptr, &safeWritePointer);
 				if (FAILED(result)) {
-					ATA_ERROR("error (" << getErrorString(result) << ") getting current write position!");
+					ATA_ERROR(getErrorString(result) << ": getting current write position!");
 					return;
 				}
 				result = dsCaptureBuffer->GetCurrentPosition(nullptr, &safeReadPointer);
 				if (FAILED(result)) {
-					ATA_ERROR("error (" << getErrorString(result) << ") getting current read position!");
+					ATA_ERROR(getErrorString(result) << ": getting current read position!");
 					return;
 				}
 				if (    safeWritePointer != startSafeWritePointer
@@ -1078,7 +1091,7 @@ void audio::orchestra::api::Ds::callbackEvent() {
 			LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) m_private->buffer[0];
 			result = dsWriteBuffer->GetCurrentPosition(&currentWritePointer, &safeWritePointer);
 			if (FAILED(result)) {
-				ATA_ERROR("error (" << getErrorString(result) << ") getting current write position!");
+				ATA_ERROR(getErrorString(result) << ": getting current write position!");
 				return;
 			}
 			m_private->bufferPointer[0] = safeWritePointer + m_private->dsPointerLeadTime[0];
@@ -1123,7 +1136,7 @@ void audio::orchestra::api::Ds::callbackEvent() {
 			// Find out where the read and "safe write" pointers are.
 			result = dsBuffer->GetCurrentPosition(&currentWritePointer, &safeWritePointer);
 			if (FAILED(result)) {
-				ATA_ERROR("error (" << getErrorString(result) << ") getting current write position!");
+				ATA_ERROR(getErrorString(result) << ": getting current write position!");
 				return;
 			}
 			// We will copy our output buffer into the region between
@@ -1172,7 +1185,7 @@ void audio::orchestra::api::Ds::callbackEvent() {
 		                        &bufferSize2,
 		                        0);
 		if (FAILED(result)) {
-			ATA_ERROR("error (" << getErrorString(result) << ") locking buffer during playback!");
+			ATA_ERROR(getErrorString(result) << ": locking buffer during playback!");
 			return;
 		}
 		// Copy our buffer into the DS buffer
@@ -1183,7 +1196,7 @@ void audio::orchestra::api::Ds::callbackEvent() {
 		// Update our buffer offset and unlock sound buffer
 		dsBuffer->Unlock(buffer1, bufferSize1, buffer2, bufferSize2);
 		if (FAILED(result)) {
-			ATA_ERROR("error (" << getErrorString(result) << ") unlocking buffer during playback!");
+			ATA_ERROR(getErrorString(result) << ": unlocking buffer during playback!");
 			return;
 		}
 		nextWritePointer = (nextWritePointer + bufferSize1 + bufferSize2) % dsBufferSize;
@@ -1211,7 +1224,7 @@ void audio::orchestra::api::Ds::callbackEvent() {
 		// Find out where the write and "safe read" pointers are.
 		result = dsBuffer->GetCurrentPosition(&currentReadPointer, &safeReadPointer);
 		if (FAILED(result)) {
-			ATA_ERROR("error (" << getErrorString(result) << ") getting current read position!");
+			ATA_ERROR(getErrorString(result) << ": getting current read position!");
 			return;
 		}
 		if (safeReadPointer < (DWORD)nextReadPointer) {
@@ -1271,7 +1284,7 @@ void audio::orchestra::api::Ds::callbackEvent() {
 				// Wake up and find out where we are now.
 				result = dsBuffer->GetCurrentPosition(&currentReadPointer, &safeReadPointer);
 				if (FAILED(result)) {
-					ATA_ERROR("error (" << getErrorString(result) << ") getting current read position!");
+					ATA_ERROR(getErrorString(result) << ": getting current read position!");
 					return;
 				}
 				if (safeReadPointer < (DWORD)nextReadPointer) {
@@ -1289,7 +1302,7 @@ void audio::orchestra::api::Ds::callbackEvent() {
 		                        &bufferSize2,
 		                        0);
 		if (FAILED(result)) {
-			ATA_ERROR("error (" << getErrorString(result) << ") locking capture buffer!");
+			ATA_ERROR(getErrorString(result) << ": locking capture buffer!");
 			return;
 		}
 		if (m_duplexPrerollBytes <= 0) {
@@ -1309,7 +1322,7 @@ void audio::orchestra::api::Ds::callbackEvent() {
 		nextReadPointer = (nextReadPointer + bufferSize1 + bufferSize2) % dsBufferSize;
 		dsBuffer->Unlock(buffer1, bufferSize1, buffer2, bufferSize2);
 		if (FAILED(result)) {
-			ATA_ERROR("error (" << getErrorString(result) << ") unlocking capture buffer!");
+			ATA_ERROR(getErrorString(result) << ": unlocking capture buffer!");
 			return;
 		}
 		m_private->bufferPointer[1] = nextReadPointer;
@@ -1330,100 +1343,12 @@ unlock:
 }
 
 void audio::orchestra::api::Ds::dsCallbackEvent(void *_userData) {
-	etk::thread::setName("DS IO-" + m_name);
 	audio::orchestra::api::Ds* myClass = reinterpret_cast<audio::orchestra::api::Ds*>(_userData);
 	while (myClass->m_private->threadRunning == true) {
 		myClass->callbackEvent();
 	}
 }
 
-#include "tchar.h"
-static std::string convertTChar(LPCTSTR _name) {
-#if defined(UNICODE) || defined(_UNICODE)
-	int32_t length = WideCharToMultiByte(CP_UTF8, 0, _name, -1, nullptr, 0, nullptr, nullptr);
-	std::string s(length-1, '\0');
-	WideCharToMultiByte(CP_UTF8, 0, _name, -1, &s[0], length, nullptr, nullptr);
-#else
-	std::string s(_name);
-#endif
-	return s;
-}
-
-static BOOL CALLBACK deviceQueryCallback(LPGUID _lpguid,
-                                         LPCTSTR _description,
-                                         LPCTSTR _module,
-                                         LPVOID _lpContext) {
-	struct DsProbeData& probeInfo = *(struct DsProbeData*) _lpContext;
-	std::vector<DsDevice>& dsDevices = *probeInfo.dsDevices;
-	HRESULT hr;
-	bool validDevice = false;
-	if (probeInfo.isInput == true) {
-		DSCCAPS caps;
-		LPDIRECTSOUNDCAPTURE object;
-		hr = DirectSoundCaptureCreate(_lpguid, &object,	 nullptr);
-		if (hr != DS_OK) {
-			return TRUE;
-		}
-		caps.dwSize = sizeof(caps);
-		hr = object->GetCaps(&caps);
-		if (hr == DS_OK) {
-			if (caps.dwChannels > 0 && caps.dwFormats > 0) {
-				validDevice = true;
-			}
-		}
-		object->Release();
-	} else {
-		DSCAPS caps;
-		LPDIRECTSOUND object;
-		hr = DirectSoundCreate(_lpguid, &object,	 nullptr);
-		if (hr != DS_OK) {
-			return TRUE;
-		}
-		caps.dwSize = sizeof(caps);
-		hr = object->GetCaps(&caps);
-		if (hr == DS_OK) {
-			if (    caps.dwFlags & DSCAPS_PRIMARYMONO
-			     || caps.dwFlags & DSCAPS_PRIMARYSTEREO) {
-				validDevice = true;
-			}
-		}
-		object->Release();
-	}
-	// If good device, then save its name and guid.
-	std::string name = convertTChar(_description);
-	//if (name == "Primary Sound Driver" || name == "Primary Sound Capture Driver")
-	if (_lpguid == nullptr) {
-		name = "Default Device";
-	}
-	if (validDevice) {
-		for (size_t i=0; i<dsDevices.size(); i++) {
-			if (dsDevices[i].name == name) {
-				dsDevices[i].found = true;
-				if (probeInfo.isInput) {
-					dsDevices[i].id[1] = _lpguid;
-					dsDevices[i].validId[1] = true;
-				} else {
-					dsDevices[i].id[0] = _lpguid;
-					dsDevices[i].validId[0] = true;
-				}
-				return TRUE;
-			}
-		}
-		DsDevice device;
-		device.name = name;
-		device.found = true;
-		if (probeInfo.isInput) {
-			device.id[1] = _lpguid;
-			device.validId[1] = true;
-		} else {
-			device.id[0] = _lpguid;
-			device.validId[0] = true;
-		}
-		dsDevices.push_back(device);
-	}
-	return TRUE;
-}
-
 static const char* getErrorString(int32_t code) {
 	switch (code) {
 		case DSERR_ALLOCATED:
diff --git a/audio/orchestra/api/Ds.h b/audio/orchestra/api/Ds.h
index e2bd8e1..9c7af8d 100644
--- a/audio/orchestra/api/Ds.h
+++ b/audio/orchestra/api/Ds.h
@@ -23,8 +23,6 @@ namespace audio {
 						return audio::orchestra::type_ds;
 					}
 					uint32_t getDeviceCount();
-					uint32_t getDefaultOutputDevice();
-					uint32_t getDefaultInputDevice();
 					audio::orchestra::DeviceInfo getDeviceInfo(uint32_t _device);
 					enum audio::orchestra::error closeStream();
 					enum audio::orchestra::error startStream();
diff --git a/audio/orchestra/api/Jack.cpp b/audio/orchestra/api/Jack.cpp
index d5bd2ec..bf62a73 100644
--- a/audio/orchestra/api/Jack.cpp
+++ b/audio/orchestra/api/Jack.cpp
@@ -125,12 +125,11 @@ uint32_t audio::orchestra::api::Jack::getDeviceCount() {
 		free(ports);
 	}
 	jack_client_close(client);
-	return nDevices;
+	return nDevices*2;
 }
 
 audio::orchestra::DeviceInfo audio::orchestra::api::Jack::getDeviceInfo(uint32_t _device) {
 	audio::orchestra::DeviceInfo info;
-	info.probed = false;
 	jack_options_t options = (jack_options_t) (JackNoStartServer); //JackNullOption
 	jack_status_t *status = nullptr;
 	jack_client_t *client = jack_client_open("orchestraJackInfo", options, status);
@@ -143,16 +142,18 @@ audio::orchestra::DeviceInfo audio::orchestra::api::Jack::getDeviceInfo(uint32_t
 	std::string port, previousPort;
 	uint32_t nPorts = 0, nDevices = 0;
 	ports = jack_get_ports(client, nullptr, nullptr, 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 ];
+			port = (char *) ports[nPorts];
 			iColon = port.find(":");
 			if (iColon != std::string::npos) {
 				port = port.substr(0, iColon);
 				if (port != previousPort) {
-					if (nDevices == _device) {
+					if (nDevices == deviceID) {
 						info.name = port;
 					}
 					nDevices++;
@@ -162,7 +163,7 @@ audio::orchestra::DeviceInfo audio::orchestra::api::Jack::getDeviceInfo(uint32_t
 		} while (ports[++nPorts]);
 		free(ports);
 	}
-	if (_device >= nDevices) {
+	if (deviceID >= nDevices) {
 		jack_client_close(client);
 		ATA_ERROR("device ID is invalid!");
 		// TODO : audio::orchestra::error_invalidUse;
@@ -171,50 +172,42 @@ audio::orchestra::DeviceInfo audio::orchestra::api::Jack::getDeviceInfo(uint32_t
 	// Get the current jack server sample rate.
 	info.sampleRates.clear();
 	info.sampleRates.push_back(jack_get_sample_rate(client));
-	// Count the available ports containing the client name as device
-	// channels.	Jack "input ports" equal RtAudio output channels.
-	uint32_t nChannels = 0;
-	ports = jack_get_ports(client, info.name.c_str(), nullptr, JackPortIsInput);
-	if (ports) {
-		while (ports[ nChannels ]) {
-			nChannels++;
+	if (info.input == true) {
+		ports = jack_get_ports(client, info.name.c_str(), nullptr, JackPortIsOutput);
+		if (ports) {
+			int32_t iii=0;
+			while (ports[iii]) {
+				ATA_ERROR(" ploppp='" << ports[iii] << "'");
+				info.channels.push_back(audio::channel_unknow);
+				iii++;
+			}
+			free(ports);
 		}
-		free(ports);
-		info.outputChannels = nChannels;
-	}
-	// Jack "output ports" equal RtAudio input channels.
-	nChannels = 0;
-	ports = jack_get_ports(client, info.name.c_str(), nullptr, JackPortIsOutput);
-	if (ports) {
-		while (ports[ nChannels ]) {
-			nChannels++;
+	} else {
+		ports = jack_get_ports(client, info.name.c_str(), nullptr, JackPortIsInput);
+		if (ports) {
+			int32_t iii=0;
+			while (ports[iii]) {
+				ATA_ERROR(" ploppp='" << ports[iii] << "'");
+				info.channels.push_back(audio::channel_unknow);
+				iii++;
+			}
+			free(ports);
 		}
-		free(ports);
-		info.inputChannels = nChannels;
 	}
-	if (info.outputChannels == 0 && info.inputChannels == 0) {
+	if (info.channels.size() == 0) {
 		jack_client_close(client);
 		ATA_ERROR("error determining Jack input/output channels!");
 		// TODO : audio::orchestra::error_warning;
 		return info;
 	}
-	// If device opens for both playback and capture, we determine the channels.
-	if (info.outputChannels > 0 && info.inputChannels > 0) {
-		info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
-	}
 	// Jack always uses 32-bit floats.
 	info.nativeFormats.push_back(audio::format_float);
 	// Jack doesn't provide default devices so we'll use the first available one.
-	if (    _device == 0
-	     && info.outputChannels > 0) {
-		info.isDefaultOutput = true;
-	}
-	if (    _device == 0
-	     && info.inputChannels > 0) {
-		info.isDefaultInput = true;
+	if (deviceID == 0) {
+		info.isDefault = true;
 	}
 	jack_client_close(client);
-	info.probed = true;
 	return info;
 }
 
@@ -295,6 +288,8 @@ bool audio::orchestra::api::Jack::probeDeviceOpen(uint32_t _device,
 	const char **ports;
 	std::string port, previousPort, deviceName;
 	uint32_t nPorts = 0, nDevices = 0;
+	int32_t deviceID = _device/2;
+	bool isInput = _device%2==0?true:false;
 	ports = jack_get_ports(client, nullptr, nullptr, 0);
 	if (ports) {
 		// Parse the port names up to the first colon (:).
@@ -305,7 +300,7 @@ bool audio::orchestra::api::Jack::probeDeviceOpen(uint32_t _device,
 			if (iColon != std::string::npos) {
 				port = port.substr(0, iColon);
 				if (port != previousPort) {
-					if (nDevices == _device) {
+					if (nDevices == deviceID) {
 						deviceName = port;
 					}
 					nDevices++;
@@ -323,7 +318,9 @@ bool audio::orchestra::api::Jack::probeDeviceOpen(uint32_t _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;
+	if (_mode == audio::orchestra::mode_input) {
+		flag = JackPortIsOutput;
+	}
 	ports = jack_get_ports(client, deviceName.c_str(), nullptr, flag);
 	if (ports) {
 		while (ports[ nChannels ]) {
diff --git a/audio/orchestra/api/Pulse.cpp b/audio/orchestra/api/Pulse.cpp
index 7e0a0f3..1cc0940 100644
--- a/audio/orchestra/api/Pulse.cpp
+++ b/audio/orchestra/api/Pulse.cpp
@@ -81,8 +81,8 @@ audio::orchestra::api::Pulse::~Pulse() {
 }
 
 uint32_t audio::orchestra::api::Pulse::getDeviceCount() {
-	#if 0
-		std::vector<audio::orchestra::api::pulse::Element> list = audio::orchestra::api::pulse::getDeviceList();
+	#if 1
+		std::vector<audio::orchestra::DeviceInfo> list = audio::orchestra::api::pulse::getDeviceList();
 		return list.size();
 	#else
 		return 1;
@@ -90,23 +90,12 @@ uint32_t audio::orchestra::api::Pulse::getDeviceCount() {
 }
 
 audio::orchestra::DeviceInfo audio::orchestra::api::Pulse::getDeviceInfo(uint32_t _device) {
-	audio::orchestra::DeviceInfo info;
-	//std::vector<audio::orchestra::api::pulse::Element> list = audio::orchestra::api::pulse::getDeviceList();
-	// TODO : Do it better... it is a little poor ...
-	info.probed = true;
-	info.name = "PulseAudio";
-	info.outputChannels = 2;
-	info.inputChannels = 2;
-	info.duplexChannels = 0;
-	info.isDefaultOutput = true;
-	info.isDefaultInput = true;
-	for (const uint32_t *sr = SUPPORTED_SAMPLERATES; *sr; ++sr) {
-		info.sampleRates.push_back(*sr);
+	std::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();
 	}
-	info.nativeFormats.push_back(audio::format_int16);
-	info.nativeFormats.push_back(audio::format_int32);
-	info.nativeFormats.push_back(audio::format_float);
-	return info;
+	return list[_device];
 }
 
 static void pulseaudio_callback(void* _userData) {
diff --git a/audio/orchestra/api/PulseDeviceList.cpp b/audio/orchestra/api/PulseDeviceList.cpp
index 4395ef0..a49235f 100644
--- a/audio/orchestra/api/PulseDeviceList.cpp
+++ b/audio/orchestra/api/PulseDeviceList.cpp
@@ -9,6 +9,10 @@
 #include <pulse/pulseaudio.h>
 #include <audio/orchestra/api/PulseDeviceList.h>
 #include <audio/orchestra/debug.h>
+#include <audio/Time.h>
+#include <audio/Duration.h>
+#include <audio/format.h>
+#include <etk/stdTools.h>
 
 // This callback gets called when our context changes state.  We really only
 // care about when it's ready or if it has failed
@@ -19,87 +23,259 @@ static void callbackStateMachine(pa_context* _contex, void *_userdata) {
 	switch  (state) {
 		// There are just here for reference
 		case PA_CONTEXT_UNCONNECTED:
-			ATA_INFO("pulse state: PA_CONTEXT_UNCONNECTED");
+			ATA_VERBOSE("pulse state: PA_CONTEXT_UNCONNECTED");
 			break;
 		case PA_CONTEXT_CONNECTING:
-			ATA_INFO("pulse state: PA_CONTEXT_CONNECTING");
+			ATA_VERBOSE("pulse state: PA_CONTEXT_CONNECTING");
 			break;
 		case PA_CONTEXT_AUTHORIZING:
-			ATA_INFO("pulse state: PA_CONTEXT_AUTHORIZING");
+			ATA_VERBOSE("pulse state: PA_CONTEXT_AUTHORIZING");
 			break;
 		case PA_CONTEXT_SETTING_NAME:
-			ATA_INFO("pulse state: PA_CONTEXT_SETTING_NAME");
+			ATA_VERBOSE("pulse state: PA_CONTEXT_SETTING_NAME");
 			break;
 		default:
-			ATA_INFO("pulse state: default");
+			ATA_VERBOSE("pulse state: default");
 			break;
 		case PA_CONTEXT_FAILED:
 			*pulseAudioReady = 2;
-			ATA_INFO("pulse state: PA_CONTEXT_FAILED");
+			ATA_VERBOSE("pulse state: PA_CONTEXT_FAILED");
 			break;
 		case PA_CONTEXT_TERMINATED:
 			*pulseAudioReady = 2;
-			ATA_INFO("pulse state: PA_CONTEXT_TERMINATED");
+			ATA_VERBOSE("pulse state: PA_CONTEXT_TERMINATED");
 			break;
 		case PA_CONTEXT_READY:
 			*pulseAudioReady = 1;
-			ATA_INFO("pulse state: PA_CONTEXT_READY");
+			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 std::vector<audio::channel> getChannelOrderFromPulseChannel(const struct pa_channel_map& _map) {
+	std::vector<audio::channel> out;
+	
+	for (int32_t iii=0; iii<_map.channels; ++iii) {
+		switch(_map.map[iii]) {
+			default:
+			case PA_CHANNEL_POSITION_MAX:
+			case PA_CHANNEL_POSITION_INVALID:
+				out.push_back(audio::channel_unknow);
+				break;
+			case PA_CHANNEL_POSITION_MONO:
+			case PA_CHANNEL_POSITION_FRONT_CENTER:
+				out.push_back(audio::channel_frontCenter);
+				break;
+			case PA_CHANNEL_POSITION_FRONT_LEFT:
+				out.push_back(audio::channel_frontLeft);
+				break;
+			case PA_CHANNEL_POSITION_FRONT_RIGHT:
+				out.push_back(audio::channel_frontRight);
+				break;
+			case PA_CHANNEL_POSITION_REAR_CENTER:
+				out.push_back(audio::channel_rearCenter);
+				break;
+			case PA_CHANNEL_POSITION_REAR_LEFT:
+				out.push_back(audio::channel_rearLeft);
+				break;
+			case PA_CHANNEL_POSITION_REAR_RIGHT:
+				out.push_back(audio::channel_rearRight);
+				break;
+			case PA_CHANNEL_POSITION_LFE:
+				out.push_back(audio::channel_lfe);
+				break;
+			case PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER:
+				out.push_back(audio::channel_centerLeft);
+				break;
+			case PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER:
+				out.push_back(audio::channel_centerRight);
+				break;
+			case PA_CHANNEL_POSITION_SIDE_LEFT:
+				out.push_back(audio::channel_topCenterLeft);
+				break;
+			case PA_CHANNEL_POSITION_SIDE_RIGHT:
+				out.push_back(audio::channel_topCenterRight);
+				break;
+			case PA_CHANNEL_POSITION_TOP_CENTER:
+			case PA_CHANNEL_POSITION_TOP_FRONT_CENTER:
+				out.push_back(audio::channel_topFrontCenter);
+				break;
+			case PA_CHANNEL_POSITION_TOP_FRONT_LEFT:
+				out.push_back(audio::channel_topFrontLeft);
+				break;
+			case PA_CHANNEL_POSITION_TOP_FRONT_RIGHT:
+				out.push_back(audio::channel_topFrontRight);
+				break;
+			case PA_CHANNEL_POSITION_TOP_REAR_LEFT:
+				out.push_back(audio::channel_topRearLeft);
+				break;
+			case PA_CHANNEL_POSITION_TOP_REAR_RIGHT:
+				out.push_back(audio::channel_topRearRight);
+				break;
+			case PA_CHANNEL_POSITION_TOP_REAR_CENTER:
+				out.push_back(audio::channel_topRearCenter);
+				break;
+			case PA_CHANNEL_POSITION_AUX0:  out.push_back(audio::channel_aux0);  break;
+			case PA_CHANNEL_POSITION_AUX1:  out.push_back(audio::channel_aux1);  break;
+			case PA_CHANNEL_POSITION_AUX2:  out.push_back(audio::channel_aux2);  break;
+			case PA_CHANNEL_POSITION_AUX3:  out.push_back(audio::channel_aux3);  break;
+			case PA_CHANNEL_POSITION_AUX4:  out.push_back(audio::channel_aux4);  break;
+			case PA_CHANNEL_POSITION_AUX5:  out.push_back(audio::channel_aux5);  break;
+			case PA_CHANNEL_POSITION_AUX6:  out.push_back(audio::channel_aux6);  break;
+			case PA_CHANNEL_POSITION_AUX7:  out.push_back(audio::channel_aux7);  break;
+			case PA_CHANNEL_POSITION_AUX8:  out.push_back(audio::channel_aux8);  break;
+			case PA_CHANNEL_POSITION_AUX9:  out.push_back(audio::channel_aux9);  break;
+			case PA_CHANNEL_POSITION_AUX10: out.push_back(audio::channel_aux10); break;
+			case PA_CHANNEL_POSITION_AUX11: out.push_back(audio::channel_aux11); break;
+			case PA_CHANNEL_POSITION_AUX12: out.push_back(audio::channel_aux12); break;
+			case PA_CHANNEL_POSITION_AUX13: out.push_back(audio::channel_aux13); break;
+			case PA_CHANNEL_POSITION_AUX14: out.push_back(audio::channel_aux14); break;
+			case PA_CHANNEL_POSITION_AUX15: out.push_back(audio::channel_aux15); break;
+			case PA_CHANNEL_POSITION_AUX16: out.push_back(audio::channel_aux16); break;
+			case PA_CHANNEL_POSITION_AUX17: out.push_back(audio::channel_aux17); break;
+			case PA_CHANNEL_POSITION_AUX18: out.push_back(audio::channel_aux18); break;
+			case PA_CHANNEL_POSITION_AUX19: out.push_back(audio::channel_aux19); break;
+			case PA_CHANNEL_POSITION_AUX20: out.push_back(audio::channel_aux20); break;
+			case PA_CHANNEL_POSITION_AUX21: out.push_back(audio::channel_aux21); break;
+			case PA_CHANNEL_POSITION_AUX22: out.push_back(audio::channel_aux22); break;
+			case PA_CHANNEL_POSITION_AUX23: out.push_back(audio::channel_aux23); break;
+			case PA_CHANNEL_POSITION_AUX24: out.push_back(audio::channel_aux24); break;
+			case PA_CHANNEL_POSITION_AUX25: out.push_back(audio::channel_aux25); break;
+			case PA_CHANNEL_POSITION_AUX26: out.push_back(audio::channel_aux26); break;
+			case PA_CHANNEL_POSITION_AUX27: out.push_back(audio::channel_aux27); break;
+			case PA_CHANNEL_POSITION_AUX28: out.push_back(audio::channel_aux28); break;
+			case PA_CHANNEL_POSITION_AUX29: out.push_back(audio::channel_aux29); break;
+			case PA_CHANNEL_POSITION_AUX30: out.push_back(audio::channel_aux30); break;
+			case PA_CHANNEL_POSITION_AUX31: out.push_back(audio::channel_aux31); break;
+		}
+	}
+	
+	return out;
+}
 // Callback on getting data from pulseaudio:
 static void callbackGetSinkList(pa_context* _contex, const pa_sink_info* _info, int _eol, void* _userdata) {
-	std::vector<audio::orchestra::api::pulse::Element>* list = static_cast<std::vector<audio::orchestra::api::pulse::Element>*>(_userdata);
+	std::vector<audio::orchestra::DeviceInfo>* list = static_cast<std::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;
 	}
-	ATA_INFO("find output : " << _info->name);
-	list->push_back(audio::orchestra::api::pulse::Element(_info->index, false, _info->name, _info->description));
+	audio::orchestra::DeviceInfo info;
+	info.input = false;
+	info.name = _info->name;
+	info.desc = _info->description;
+	info.sampleRates.push_back(_info->sample_spec.rate);
+	info.nativeFormats.push_back(getFormatFromPulseFormat(_info->sample_spec.format));
+	info.channels = getChannelOrderFromPulseChannel(_info->channel_map);
+	ATA_VERBOSE("plop=" << _info->index << " " << _info->name);
+	//ATA_DEBUG("          ports=" << _info->n_ports);
+	list->push_back(info);
 }
 
 // allback to get data from pulseaudio:
 static void callbackGetSourceList(pa_context* _contex, const pa_source_info* _info, int _eol, void* _userdata) {
-	std::vector<audio::orchestra::api::pulse::Element>* list = static_cast<std::vector<audio::orchestra::api::pulse::Element>*>(_userdata);
+	std::vector<audio::orchestra::DeviceInfo>* list = static_cast<std::vector<audio::orchestra::DeviceInfo>*>(_userdata);
 	if (_eol > 0) {
 		return;
 	}
-	ATA_INFO("find input : " << _info->name);
-	list->push_back(audio::orchestra::api::pulse::Element(_info->index, true, _info->name, _info->description));
+	audio::orchestra::DeviceInfo info;
+	info.input = true;
+	info.name = _info->name;
+	info.desc = _info->description;
+	info.sampleRates.push_back(_info->sample_spec.rate);
+	info.nativeFormats.push_back(getFormatFromPulseFormat(_info->sample_spec.format));
+	info.channels = getChannelOrderFromPulseChannel(_info->channel_map);
+	ATA_VERBOSE("plop=" << _info->index << " " << _info->name);
+	list->push_back(info);
 }
 
-std::vector<audio::orchestra::api::pulse::Element> audio::orchestra::api::pulse::getDeviceList() {
+// to not update all the time ...
+static std::vector<audio::orchestra::DeviceInfo> pulseAudioListOfDevice;
+static audio::Time pulseAudioListOfDeviceTime;
+
+std::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;
-	std::vector<audio::orchestra::api::pulse::Element> out;
+	pa_context_flags_t pulseAudioFlags = PA_CONTEXT_NOAUTOSPAWN;
+	std::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, "test");
-	// This function connects to the pulse server
-	pa_context_connect(pulseAudioContex, NULL, pulseAudioFlags, NULL);
+	pulseAudioContex = pa_context_new(pulseAudioMainLoopAPI, "orchestraPulseCount");
 	// If there's an error, the callback will set pulseAudioReady
 	pa_context_set_state_callback(pulseAudioContex, callbackStateMachine, &pulseAudioReady);
-	ATA_INFO("start main loop...");
-	while (true) {
-		ATA_INFO("loop");
+	// 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) {
-			ATA_INFO("Pulse not ready");
 			pa_mainloop_iterate(pulseAudioMainLoop, 1, nullptr);
 			continue;
 		}
 		// We couldn't get a connection to the server, so exit out
 		if (pulseAudioReady == 2) {
-			ATA_INFO("pulse not ready");
 			pa_context_disconnect(pulseAudioContex);
 			pa_context_unref(pulseAudioContex);
 			pa_mainloop_free(pulseAudioMainLoop);
@@ -111,7 +287,7 @@ std::vector<audio::orchestra::api::pulse::Element> audio::orchestra::api::pulse:
 		switch (state) {
 			// State 0: we haven't done anything yet
 			case 0:
-				ATA_INFO("Request sink list");
+				ATA_DEBUG("Request sink list");
 				pulseAudioOperation = pa_context_get_sink_info_list(pulseAudioContex,
 				                                                    callbackGetSinkList,
 				                                                    &out);
@@ -123,7 +299,7 @@ std::vector<audio::orchestra::api::pulse::Element> audio::orchestra::api::pulse:
 				// along to the next state
 				if (pa_operation_get_state(pulseAudioOperation) == PA_OPERATION_DONE) {
 					pa_operation_unref(pulseAudioOperation);
-					ATA_INFO("Request sources list");
+					ATA_DEBUG("Request sources list");
 					pulseAudioOperation = pa_context_get_source_info_list(pulseAudioContex,
 					                                                      callbackGetSourceList,
 					                                                      &out);
@@ -132,13 +308,14 @@ std::vector<audio::orchestra::api::pulse::Element> audio::orchestra::api::pulse:
 				break;
 			case 2:
 				if (pa_operation_get_state(pulseAudioOperation) == PA_OPERATION_DONE) {
-					ATA_INFO("All is 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);
-					return out;
+					playLoop = false;
+					break;
 				}
 			break;
 			default:
@@ -147,7 +324,32 @@ std::vector<audio::orchestra::api::pulse::Element> audio::orchestra::api::pulse:
 				return out;
 		}
 		// Iterate the main loop ..
-		pa_mainloop_iterate(pulseAudioMainLoop, 1, nullptr);
+		if (playLoop == true) {
+			pa_mainloop_iterate(pulseAudioMainLoop, 1, nullptr);
+		}
+	}
+	// 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;
 }
diff --git a/audio/orchestra/api/PulseDeviceList.h b/audio/orchestra/api/PulseDeviceList.h
index 53d84cb..e70d029 100644
--- a/audio/orchestra/api/PulseDeviceList.h
+++ b/audio/orchestra/api/PulseDeviceList.h
@@ -9,39 +9,13 @@
 #define __AUDIO_ORCHESTRA_API_PULSE_DEVICE_H__
 
 #include <etk/types.h>
+#include <audio/orchestra/DeviceInfo.h>
 
 namespace audio {
 	namespace orchestra {
 		namespace api {
 			namespace pulse {
-				class Element {
-					private:
-						size_t m_index;
-						bool m_input;
-						std::string m_name;
-						std::string m_description;
-					public:
-						Element(size_t _index, bool _input, const std::string& _name, const std::string& _desc) :
-						  m_index(_index),
-						  m_input(_input),
-						  m_name(_name),
-						  m_description(_desc) {
-							// nothing to do...
-						}
-						size_t getIndex() const {
-							return m_index;
-						}
-						bool isInput() const {
-							return m_input;
-						}
-						const std::string& getName() const {
-							return m_name;
-						}
-						const std::string& getDescription() const {
-							return m_description;
-						}
-				};
-				std::vector<audio::orchestra::api::pulse::Element> getDeviceList();
+				std::vector<audio::orchestra::DeviceInfo> getDeviceList();
 			}
 		}
 	}