[hamradio-commits] [soapyaudio] 01/02: Imported Upstream version 0~git20160607
Andreas E. Bombe
aeb at moszumanska.debian.org
Thu Aug 25 01:25:32 UTC 2016
This is an automated email from the git hooks/post-receive script.
aeb pushed a commit to branch master
in repository soapyaudio.
commit 57702b030bdf02cf1278caa51fce9aa1dc522a8d
Author: Andreas Bombe <aeb at debian.org>
Date: Wed Aug 24 21:57:41 2016 +0200
Imported Upstream version 0~git20160607
---
.gitignore | 1 +
CMakeLists.txt | 158 +
Findhamlib.cmake | 59 +
LICENSE.txt | 21 +
LibFindMacros.cmake | 99 +
README.md | 11 +
Registation.cpp | 112 +
RigThread.cpp | 101 +
RigThread.h | 49 +
RtAudio/FunctionDiscoveryKeys_devpkey.h | 212 +
RtAudio/RtAudio.cpp | 10229 ++++++++++++++++++++++++++++++
RtAudio/RtAudio.h | 1163 ++++
RtAudio/readme | 61 +
Settings.cpp | 603 ++
SoapyAudio.hpp | 275 +
Streaming.cpp | 678 ++
debian/changelog | 5 +
debian/compat | 1 +
debian/control | 22 +
debian/copyright | 50 +
debian/docs | 1 +
debian/rules | 20 +
debian/source/format | 1 +
23 files changed, 13932 insertions(+)
diff --git a/.gitignore b/.gitignore
new file mode 100644
index 0000000..378eac2
--- /dev/null
+++ b/.gitignore
@@ -0,0 +1 @@
+build
diff --git a/CMakeLists.txt b/CMakeLists.txt
new file mode 100644
index 0000000..66b1cdb
--- /dev/null
+++ b/CMakeLists.txt
@@ -0,0 +1,158 @@
+########################################################################
+# Build Soapy SDR support module for Audio Devices
+########################################################################
+cmake_minimum_required(VERSION 2.8.7)
+project(SoapyAudio CXX)
+
+find_package(SoapySDR "0.4.0" NO_MODULE REQUIRED)
+if (NOT SoapySDR_FOUND)
+ message(FATAL_ERROR "Soapy SDR development files not found...")
+endif ()
+
+list(APPEND CMAKE_MODULE_PATH ${CMAKE_CURRENT_SOURCE_DIR})
+
+option(USE_HAMLIB OFF "Support hamlib for radio control functions.")
+
+if (USE_HAMLIB)
+ find_package(hamlib REQUIRED)
+
+ if (NOT hamlib_FOUND)
+ message(FATAL_ERROR "hamlib development files not found...")
+ endif ()
+
+ include_directories(${hamlib_INCLUDE_DIRS})
+ if (${hamlib_STATIC_FOUND})
+ link_libraries(${hamlib_STATIC_LIBRARIES})
+ else()
+ link_libraries(${hamlib_LIBRARIES})
+ endif()
+
+ ADD_DEFINITIONS(-DUSE_HAMLIB)
+endif ()
+
+# list(APPEND CMAKE_MODULE_PATH ${CMAKE_CURRENT_SOURCE_DIR})
+
+include_directories(${CMAKE_CURRENT_SOURCE_DIR})
+include_directories(${CMAKE_CURRENT_SOURCE_DIR}/RtAudio)
+
+#enable c++11 features
+if(CMAKE_COMPILER_IS_GNUCXX)
+
+ #C++11 is a required language feature for this project
+ include(CheckCXXCompilerFlag)
+ CHECK_CXX_COMPILER_FLAG("-std=c++11" HAS_STD_CXX11)
+ if(HAS_STD_CXX11)
+ set(CMAKE_CXX_FLAGS "${CMAKE_CXX_FLAGS} -std=c++11")
+ else(HAS_STD_CXX11)
+ set(CMAKE_CXX_FLAGS "${CMAKE_CXX_FLAGS} -std=c++0x")
+ endif()
+
+ #Thread support enabled (not the same as -lpthread)
+ list(APPEND AUDIO_LIBS -pthread)
+
+ #disable warnings for unused parameters
+ add_definitions(-Wno-unused-parameter)
+
+endif(CMAKE_COMPILER_IS_GNUCXX)
+
+if (APPLE)
+ set(CMAKE_CXX_FLAGS "${CMAKE_CXX_FLAGS} -std=c++11 -Wc++11-extensions")
+endif(APPLE)
+
+IF (WIN32)
+ set(USE_AUDIO_DS ON CACHE BOOL "Support DirectSound Audio")
+ set(USE_AUDIO_WASAPI OFF CACHE BOOL "Support WASAPI Audio")
+ # TODO:
+ # set(USE_AUDIO_ASIO OFF CACHE BOOL "Include support for ASIO Audio")
+
+ # WASAPI
+ IF(USE_AUDIO_WASAPI)
+ ADD_DEFINITIONS(-D__WINDOWS_WASAPI__)
+ IF (NOT MSVC)
+ SET(AUDIO_LIBS ${AUDIO_LIBS} -luuid -lksuser)
+ ENDIF(NOT MSVC)
+ ENDIF(USE_AUDIO_WASAPI)
+
+ # DirectSound
+ IF (USE_AUDIO_DS)
+ ADD_DEFINITIONS(-D__WINDOWS_DS__)
+ IF (MSVC)
+ SET(AUDIO_LIBS ${AUDIO_LIBS} dsound.lib)
+ ELSE (MSVC)
+ SET(AUDIO_LIBS ${AUDIO_LIBS} -ldsound)
+ ENDIF (MSVC)
+ ENDIF(USE_AUDIO_DS)
+ENDIF (WIN32)
+
+IF (UNIX AND NOT APPLE)
+ SET(USE_AUDIO_PULSE ON CACHE BOOL "Support Pulse Audio")
+ SET(USE_AUDIO_JACK OFF CACHE BOOL "Support Jack Audio")
+ SET(USE_AUDIO_ALSA OFF CACHE BOOL "Support ALSA Audio")
+ SET(USE_AUDIO_OSS OFF CACHE BOOL "Support OSS Audio")
+
+ IF(USE_AUDIO_PULSE)
+ SET (AUDIO_LIBS ${AUDIO_LIBS} pulse-simple pulse)
+ ADD_DEFINITIONS(
+ -D__LINUX_PULSE__
+ )
+ ENDIF(USE_AUDIO_PULSE)
+
+ IF(USE_AUDIO_JACK)
+ find_package(Jack)
+ SET (AUDIO_LIBS ${AUDIO_LIBS} ${JACK_LIBRARIES})
+ ADD_DEFINITIONS(
+ -D__UNIX_JACK__
+ )
+ include_directories(${JACK_INCLUDE_DIRS})
+ ENDIF(USE_AUDIO_JACK)
+
+ IF(USE_AUDIO_ALSA)
+ SET (AUDIO_LIBS ${AUDIO_LIBS} asound)
+ set(ALSA_INCLUDE_DIR "/usr/include" CACHE FILEPATH "ALSA include path")
+ include_directories(${ALSA_INCLUDE_DIR})
+ set(ALSA_LIB_DIR "/usr/lib" CACHE FILEPATH "ALSA lib path")
+ link_directories(${ALSA_LIB_DIR})
+ ADD_DEFINITIONS(
+ -D__LINUX_ALSA__
+ )
+ ENDIF(USE_AUDIO_ALSA)
+
+ IF(USE_AUDIO_OSS)
+ SET (AUDIO_LIBS ${AUDIO_LIBS} oss)
+ ADD_DEFINITIONS(
+ -D__LINUX_OSS__
+ )
+ ENDIF(USE_AUDIO_OSS)
+ENDIF(UNIX AND NOT APPLE)
+
+IF (APPLE)
+ ADD_DEFINITIONS(
+ -D__MACOSX_CORE__
+ )
+
+FIND_LIBRARY(COREAUDIO_LIBRARY CoreAudio)
+FIND_LIBRARY(COREFOUNDATION_LIBRARY CoreFoundation)
+SET (AUDIO_LIBS ${COREAUDIO_LIBRARY} ${COREFOUNDATION_LIBRARY} ${AUDIO_LIBS} )
+ENDIF (APPLE)
+
+IF (USE_HAMLIB)
+ SET (
+ HAMLIB_SOURCES
+ RigThread.cpp
+ RigThread.h
+ )
+ENDIF()
+
+SOAPY_SDR_MODULE_UTIL(
+ TARGET audioSupport
+ SOURCES
+ SoapyAudio.hpp
+ Registation.cpp
+ Settings.cpp
+ Streaming.cpp
+ RtAudio/RtAudio.cpp
+ RtAudio/RtAudio.h
+ ${HAMLIB_SOURCES}
+ LIBRARIES
+ ${AUDIO_LIBS}
+)
diff --git a/Findhamlib.cmake b/Findhamlib.cmake
new file mode 100644
index 0000000..abed74b
--- /dev/null
+++ b/Findhamlib.cmake
@@ -0,0 +1,59 @@
+# - Try to find hamlib
+# Once done, this will define:
+#
+# hamlib_FOUND - system has Hamlib-2
+# hamlib_INCLUDE_DIRS - the Hamlib-2 include directories
+# hamlib_LIBRARIES - link these to use Hamlib-2
+# hamlib_STATIC_FOUND - system has Hamlib-2 static archive
+# hamlib_STATIC_LIBRARIES - link these to use Hamlib-2 static archive
+
+include (LibFindMacros)
+
+# pkg-config?
+find_path (__hamlib_pc_path NAMES hamlib.pc
+ PATH_SUFFIXES lib/pkgconfig
+)
+if (__hamlib_pc_path)
+ set (ENV{PKG_CONFIG_PATH} "${__hamlib_pc_path}" "$ENV{PKG_CONFIG_PATH}")
+ unset (__hamlib_pc_path CACHE)
+endif ()
+
+# Use pkg-config to get hints about paths, libs and, flags
+unset (__pkg_config_checked_hamlib CACHE)
+libfind_pkg_check_modules (PC_HAMLIB hamlib)
+
+if (NOT PC_HAMLIB_STATIC_LIBRARIES)
+ if (WIN32)
+ set (PC_HAMLIB_STATIC_LIBRARIES hamlib ws2_32)
+ else ()
+ set (PC_HAMLIB_STATIC_LIBRARIES hamlib m dl usb)
+ endif ()
+endif ()
+
+# The libraries
+libfind_library (hamlib hamlib)
+libfind_library (hamlib_STATIC libhamlib.a)
+
+find_path (hamlib_INCLUDE_DIR hamlib/rig.h)
+
+# Set the include dir variables and the libraries and let libfind_process do the rest
+set (hamlib_PROCESS_INCLUDES hamlib_INCLUDE_DIR)
+set (hamlib_PROCESS_LIBS hamlib_LIBRARY)
+libfind_process (hamlib)
+
+set (hamlib_STATIC_PROCESS_INCLUDES hamlib_STATIC_INCLUDE_DIR)
+set (hamlib_STATIC_PROCESS_LIBS hamlib_STATIC_LIBRARY PC_HAMLIB_STATIC_LIBRARIES)
+libfind_process (hamlib_STATIC)
+
+# make sure we return a full path for the library we return
+if (hamlib_FOUND)
+ list (REMOVE_ITEM hamlib_LIBRARIES hamlib)
+ if (hamlib_STATIC_LIBRARIES)
+ list (REMOVE_ITEM hamlib_STATIC_LIBRARIES hamlib)
+ endif ()
+endif ()
+
+# Handle the QUIETLY and REQUIRED arguments and set HAMLIB_FOUND to
+# TRUE if all listed variables are TRUE
+include (FindPackageHandleStandardArgs)
+find_package_handle_standard_args (hamlib DEFAULT_MSG hamlib_INCLUDE_DIRS hamlib_LIBRARY hamlib_LIBRARIES)
diff --git a/LICENSE.txt b/LICENSE.txt
new file mode 100644
index 0000000..439cd08
--- /dev/null
+++ b/LICENSE.txt
@@ -0,0 +1,21 @@
+The MIT License (MIT)
+
+Copyright (c) 2015 Charles J. Cliffe
+
+Permission is hereby granted, free of charge, to any person obtaining a copy
+of this software and associated documentation files (the "Software"), to deal
+in the Software without restriction, including without limitation the rights
+to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+copies of the Software, and to permit persons to whom the Software is
+furnished to do so, subject to the following conditions:
+
+The above copyright notice and this permission notice shall be included in
+all copies or substantial portions of the Software.
+
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+THE SOFTWARE.
\ No newline at end of file
diff --git a/LibFindMacros.cmake b/LibFindMacros.cmake
new file mode 100644
index 0000000..69975c5
--- /dev/null
+++ b/LibFindMacros.cmake
@@ -0,0 +1,99 @@
+# Works the same as find_package, but forwards the "REQUIRED" and "QUIET" arguments
+# used for the current package. For this to work, the first parameter must be the
+# prefix of the current package, then the prefix of the new package etc, which are
+# passed to find_package.
+macro (libfind_package PREFIX)
+ set (LIBFIND_PACKAGE_ARGS ${ARGN})
+ if (${PREFIX}_FIND_QUIETLY)
+ set (LIBFIND_PACKAGE_ARGS ${LIBFIND_PACKAGE_ARGS} QUIET)
+ endif (${PREFIX}_FIND_QUIETLY)
+ if (${PREFIX}_FIND_REQUIRED)
+ set (LIBFIND_PACKAGE_ARGS ${LIBFIND_PACKAGE_ARGS} REQUIRED)
+ endif (${PREFIX}_FIND_REQUIRED)
+ find_package(${LIBFIND_PACKAGE_ARGS})
+endmacro (libfind_package)
+
+# CMake developers made the UsePkgConfig system deprecated in the same release (2.6)
+# where they added pkg_check_modules. Consequently I need to support both in my scripts
+# to avoid those deprecated warnings. Here's a helper that does just that.
+# Works identically to pkg_check_modules, except that no checks are needed prior to use.
+macro (libfind_pkg_check_modules PREFIX PKGNAME)
+ if (${CMAKE_MAJOR_VERSION} EQUAL 2 AND ${CMAKE_MINOR_VERSION} EQUAL 4)
+ include(UsePkgConfig)
+ pkgconfig(${PKGNAME} ${PREFIX}_INCLUDE_DIRS ${PREFIX}_LIBRARY_DIRS ${PREFIX}_LDFLAGS ${PREFIX}_CFLAGS)
+ else (${CMAKE_MAJOR_VERSION} EQUAL 2 AND ${CMAKE_MINOR_VERSION} EQUAL 4)
+ find_package(PkgConfig)
+ if (PKG_CONFIG_FOUND)
+ pkg_check_modules(${PREFIX} ${PKGNAME})
+ endif (PKG_CONFIG_FOUND)
+ endif (${CMAKE_MAJOR_VERSION} EQUAL 2 AND ${CMAKE_MINOR_VERSION} EQUAL 4)
+endmacro (libfind_pkg_check_modules)
+
+# Do the final processing once the paths have been detected.
+# If include dirs are needed, ${PREFIX}_PROCESS_INCLUDES should be set to contain
+# all the variables, each of which contain one include directory.
+# Ditto for ${PREFIX}_PROCESS_LIBS and library files.
+# Will set ${PREFIX}_FOUND, ${PREFIX}_INCLUDE_DIRS and ${PREFIX}_LIBRARIES.
+# Also handles errors in case library detection was required, etc.
+macro (libfind_process PREFIX)
+ # Skip processing if already processed during this run
+ if (NOT ${PREFIX}_FOUND)
+ # Start with the assumption that the library was found
+ set (${PREFIX}_FOUND TRUE)
+
+ # Process all includes and set _FOUND to false if any are missing
+ foreach (i ${${PREFIX}_PROCESS_INCLUDES})
+ if (${i})
+ set (${PREFIX}_INCLUDE_DIRS ${${PREFIX}_INCLUDE_DIRS} ${${i}})
+ mark_as_advanced(${i})
+ else (${i})
+ set (${PREFIX}_FOUND FALSE)
+ endif (${i})
+ endforeach (i)
+
+ # Process all libraries and set _FOUND to false if any are missing
+ foreach (i ${${PREFIX}_PROCESS_LIBS})
+ if (${i})
+ set (${PREFIX}_LIBRARIES ${${PREFIX}_LIBRARIES} ${${i}})
+ mark_as_advanced(${i})
+ else (${i})
+ set (${PREFIX}_FOUND FALSE)
+ endif (${i})
+ endforeach (i)
+
+ # Print message and/or exit on fatal error
+ if (${PREFIX}_FOUND)
+ if (NOT ${PREFIX}_FIND_QUIETLY)
+ message (STATUS "Found ${PREFIX} ${${PREFIX}_VERSION}")
+ endif (NOT ${PREFIX}_FIND_QUIETLY)
+ else (${PREFIX}_FOUND)
+ if (${PREFIX}_FIND_REQUIRED)
+ foreach (i ${${PREFIX}_PROCESS_INCLUDES} ${${PREFIX}_PROCESS_LIBS})
+ message("${i}=${${i}}")
+ endforeach (i)
+ message (FATAL_ERROR "Required library ${PREFIX} NOT FOUND.\nInstall the library (dev version) and try again. If the library is already installed, use ccmake to set the missing variables manually.")
+ endif (${PREFIX}_FIND_REQUIRED)
+ endif (${PREFIX}_FOUND)
+ endif (NOT ${PREFIX}_FOUND)
+endmacro (libfind_process)
+
+macro(libfind_library PREFIX basename)
+ set(TMP "")
+ if(MSVC80)
+ set(TMP -vc80)
+ endif(MSVC80)
+ if(MSVC90)
+ set(TMP -vc90)
+ endif(MSVC90)
+ set(${PREFIX}_LIBNAMES ${basename}${TMP})
+ if(${ARGC} GREATER 2)
+ set(${PREFIX}_LIBNAMES ${basename}${TMP}-${ARGV2})
+ string(REGEX REPLACE "\\." "_" TMP ${${PREFIX}_LIBNAMES})
+ set(${PREFIX}_LIBNAMES ${${PREFIX}_LIBNAMES} ${TMP})
+ endif(${ARGC} GREATER 2)
+ find_library(${PREFIX}_LIBRARY
+ NAMES ${${PREFIX}_LIBNAMES}
+ PATHS ${${PREFIX}_PKGCONF_LIBRARY_DIRS}
+ )
+endmacro(libfind_library)
+
diff --git a/README.md b/README.md
new file mode 100644
index 0000000..57e81b3
--- /dev/null
+++ b/README.md
@@ -0,0 +1,11 @@
+# Soapy SDR plugin for Audio devices
+
+##Dependencies
+
+* SoapySDR - https://github.com/pothosware/SoapySDR/wiki
+* rtaudio - https://www.music.mcgill.ca/~gary/rtaudio/
+* hamlib - http://sourceforge.net/projects/hamlib/
+
+##Documentation
+
+* https://github.com/pothosware/SoapyAudio/wiki
diff --git a/Registation.cpp b/Registation.cpp
new file mode 100644
index 0000000..82a3452
--- /dev/null
+++ b/Registation.cpp
@@ -0,0 +1,112 @@
+/*
+ * The MIT License (MIT)
+ *
+ * Copyright (c) 2015 Charles J. Cliffe
+
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+#include "SoapyAudio.hpp"
+#include <SoapySDR/Registry.hpp>
+#include <cstdlib> //malloc
+
+static std::vector<SoapySDR::Kwargs> findAudio(const SoapySDR::Kwargs &args)
+{
+ std::vector<SoapySDR::Kwargs> results;
+
+ RtAudio endac;
+
+ int numDevices = endac.getDeviceCount();
+
+ for (int i = 0; i < numDevices; i++) {
+ RtAudio::DeviceInfo info = endac.getDeviceInfo(i);
+ SoapySDR::Kwargs soapyInfo;
+
+ soapyInfo["device_id"] = std::to_string(i);
+ soapyInfo["label"] = info.name;
+ soapyInfo["default_output"] = info.isDefaultOutput ? "True" : "False";
+ soapyInfo["default_input"] = info.isDefaultInput ? "True" : "False";
+
+ // std::cout << "\tInput channels: " << info.inputChannels << std::endl;
+ // std::cout << "\tOutput channels: " << info.outputChannels << std::endl;
+ // std::cout << "\tDuplex channels: " << info.duplexChannels << std::endl;
+
+ // std::cout << "\t" << "Native formats:" << std::endl;
+ // RtAudioFormat nFormats = info.nativeFormats;
+ // if (nFormats & RTAUDIO_SINT8) {
+ // std::cout << "\t\t8-bit signed integer." << std::endl;
+ // }
+ // if (nFormats & RTAUDIO_SINT16) {
+ // std::cout << "\t\t16-bit signed integer." << std::endl;
+ // }
+ // if (nFormats & RTAUDIO_SINT24) {
+ // std::cout << "\t\t24-bit signed integer." << std::endl;
+ // }
+ // if (nFormats & RTAUDIO_SINT32) {
+ // std::cout << "\t\t32-bit signed integer." << std::endl;
+ // }
+ // if (nFormats & RTAUDIO_FLOAT32) {
+ // std::cout << "\t\t32-bit float normalized between plus/minus 1.0." << std::endl;
+ // }
+ // if (nFormats & RTAUDIO_FLOAT64) {
+ // std::cout << "\t\t32-bit float normalized between plus/minus 1.0." << std::endl;
+ // }
+
+ // filtering
+ if (info.inputChannels == 0) { // filter output devices for now
+ continue;
+ }
+
+ if (args.count("device_id") != 0)
+ {
+ if (args.at("device_id") != soapyInfo.at("device_id"))
+ {
+ continue;
+ }
+ SoapySDR_logf(SOAPY_SDR_DEBUG, "Found device by device_id %s", soapyInfo.at("device_id").c_str());
+ }
+
+ results.push_back(soapyInfo);
+ }
+
+#ifdef USE_HAMLIB
+ rig_set_debug(RIG_DEBUG_ERR);
+ rig_load_all_backends();
+ SoapyAudio::rigCaps.clear();
+ rig_list_foreach(SoapyAudio::add_hamlib_rig, 0);
+ std::sort(SoapyAudio::rigCaps.begin(), SoapyAudio::rigCaps.end(), rigGreater());
+#endif
+
+ return results;
+}
+
+#ifdef USE_HAMLIB
+int SoapyAudio::add_hamlib_rig(const struct rig_caps *rc, void* f)
+{
+ rigCaps.push_back(rc);
+ return 1;
+}
+#endif
+
+static SoapySDR::Device *makeAudio(const SoapySDR::Kwargs &args)
+{
+ return new SoapyAudio(args);
+}
+
+static SoapySDR::Registry registerAudio("audio", &findAudio, &makeAudio, SOAPY_SDR_ABI_VERSION);
diff --git a/RigThread.cpp b/RigThread.cpp
new file mode 100644
index 0000000..ac96cc2
--- /dev/null
+++ b/RigThread.cpp
@@ -0,0 +1,101 @@
+#include "SoapyAudio.hpp"
+
+#ifdef USE_HAMLIB
+RigThread::RigThread() {
+ terminated.store(true);
+}
+
+RigThread::~RigThread() {
+
+}
+
+#ifdef __APPLE__
+void *RigThread::threadMain() {
+ terminated.store(false);
+ run();
+ return this;
+};
+
+void *RigThread::pthread_helper(void *context) {
+ return ((RigThread *) context)->threadMain();
+};
+#else
+void RigThread::threadMain() {
+ terminated.store(false);
+ run();
+};
+#endif
+
+void RigThread::setup(rig_model_t rig_model, std::string rig_file, int serial_rate) {
+ rigModel = rig_model;
+ rigFile = rig_file;
+ serialRate = serial_rate;
+};
+
+void RigThread::run() {
+ int retcode, status;
+
+ SoapySDR_log(SOAPY_SDR_DEBUG, "Rig thread starting.");
+
+ rig = rig_init(rigModel);
+ strncpy(rig->state.rigport.pathname, rigFile.c_str(), FILPATHLEN - 1);
+ rig->state.rigport.parm.serial.rate = serialRate;
+ retcode = rig_open(rig);
+
+ if (retcode != 0) {
+ SoapySDR_log(SOAPY_SDR_ERROR, "Rig failed to init.");
+ terminated.store(true);
+ return;
+ }
+
+ char *info_buf = (char *)rig_get_info(rig);
+
+ if (info_buf != nullptr) {
+ SoapySDR_logf(SOAPY_SDR_DEBUG, "Rig Info: %s", info_buf);
+ }
+
+ while (!terminated.load()) {
+ std::this_thread::sleep_for(std::chrono::milliseconds(150));
+ if (freqChanged.load()) {
+ status = rig_get_freq(rig, RIG_VFO_CURR, &freq);
+ if (freq != newFreq) {
+ freq = newFreq;
+ rig_set_freq(rig, RIG_VFO_CURR, freq);
+ SoapySDR_logf(SOAPY_SDR_DEBUG, "Set Rig Freq: %f", newFreq);
+ }
+
+ freqChanged.store(false);
+ } else {
+ status = rig_get_freq(rig, RIG_VFO_CURR, &freq);
+ }
+
+ SoapySDR_logf(SOAPY_SDR_DEBUG, "Rig Freq: %f", freq);
+ }
+
+ rig_close(rig);
+ rig_cleanup(rig);
+
+ SoapySDR_log(SOAPY_SDR_DEBUG, "Rig thread exiting.");
+};
+
+freq_t RigThread::getFrequency() {
+ if (freqChanged.load()) {
+ return newFreq;
+ } else {
+ return freq;
+ }
+}
+
+void RigThread::setFrequency(freq_t new_freq) {
+ newFreq = new_freq;
+ freqChanged.store(true);
+}
+
+void RigThread::terminate() {
+ terminated.store(true);
+};
+
+bool RigThread::isTerminated() {
+ return terminated.load();
+}
+#endif
\ No newline at end of file
diff --git a/RigThread.h b/RigThread.h
new file mode 100644
index 0000000..a0d39a4
--- /dev/null
+++ b/RigThread.h
@@ -0,0 +1,49 @@
+#pragma once
+
+#include <hamlib/rig.h>
+#include <hamlib/riglist.h>
+
+#ifdef USE_HAMLIB
+struct rigGreater
+{
+ bool operator()( const struct rig_caps *lx, const struct rig_caps *rx ) const {
+ std::string ln(std::string(std::string(lx->mfg_name) + " " + std::string(lx->model_name)));
+ std::string rn(std::string(std::string(rx->mfg_name) + " " + std::string(rx->model_name)));
+ return ln.compare(rn)<0;
+ }
+};
+
+class RigThread {
+public:
+ RigThread();
+ ~RigThread();
+
+ void *pthread_helper(void *context);
+
+#ifdef __APPLE__
+ void *threadMain();
+#else
+ void threadMain();
+#endif
+
+ void setup(rig_model_t rig_model, std::string rig_file, int serial_rate);
+ void run();
+
+ void terminate();
+ bool isTerminated();
+
+ freq_t getFrequency();
+ void setFrequency(freq_t new_freq);
+
+private:
+ RIG *rig;
+ rig_model_t rigModel;
+ std::string rigFile;
+ int serialRate;
+
+ freq_t freq;
+ freq_t newFreq;
+ std::atomic_bool terminated, freqChanged;
+};
+
+#endif
\ No newline at end of file
diff --git a/RtAudio/FunctionDiscoveryKeys_devpkey.h b/RtAudio/FunctionDiscoveryKeys_devpkey.h
new file mode 100644
index 0000000..854244d
--- /dev/null
+++ b/RtAudio/FunctionDiscoveryKeys_devpkey.h
@@ -0,0 +1,212 @@
+#pragma once
+
+/*++
+
+Copyright (c) Microsoft Corporation. All rights reserved.
+
+Module Name:
+
+ devpkey.h
+
+Abstract:
+
+ Defines property keys for the Plug and Play Device Property API.
+
+Author:
+
+ Jim Cavalaris (jamesca) 10-14-2003
+
+Environment:
+
+ User-mode only.
+
+Revision History:
+
+ 14-October-2003 jamesca
+
+ Creation and initial implementation.
+
+ 20-June-2006 dougb
+
+ Copied Jim's version replaced "DEFINE_DEVPROPKEY(DEVPKEY_" with "DEFINE_PROPERTYKEY(PKEY_"
+
+--*/
+
+//#include <devpropdef.h>
+
+//
+// _NAME
+//
+
+DEFINE_PROPERTYKEY(PKEY_NAME, 0xb725f130, 0x47ef, 0x101a, 0xa5, 0xf1, 0x02, 0x60, 0x8c, 0x9e, 0xeb, 0xac, 10); // DEVPROP_TYPE_STRING
+
+//
+// Device properties
+// These PKEYs correspond to the old setupapi SPDRP_XXX properties
+//
+DEFINE_PROPERTYKEY(PKEY_Device_DeviceDesc, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 2); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_Device_HardwareIds, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 3); // DEVPROP_TYPE_STRING_LIST
+DEFINE_PROPERTYKEY(PKEY_Device_CompatibleIds, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 4); // DEVPROP_TYPE_STRING_LIST
+DEFINE_PROPERTYKEY(PKEY_Device_Service, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 6); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_Device_Class, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 9); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_Device_ClassGuid, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 10); // DEVPROP_TYPE_GUID
+DEFINE_PROPERTYKEY(PKEY_Device_Driver, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 11); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_Device_ConfigFlags, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 12); // DEVPROP_TYPE_UINT32
+DEFINE_PROPERTYKEY(PKEY_Device_Manufacturer, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 13); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_Device_FriendlyName, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 14); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_Device_LocationInfo, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 15); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_Device_PDOName, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 16); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_Device_Capabilities, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 17); // DEVPROP_TYPE_UNINT32
+DEFINE_PROPERTYKEY(PKEY_Device_UINumber, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 18); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_Device_UpperFilters, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 19); // DEVPROP_TYPE_STRING_LIST
+DEFINE_PROPERTYKEY(PKEY_Device_LowerFilters, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 20); // DEVPROP_TYPE_STRING_LIST
+DEFINE_PROPERTYKEY(PKEY_Device_BusTypeGuid, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 21); // DEVPROP_TYPE_GUID
+DEFINE_PROPERTYKEY(PKEY_Device_LegacyBusType, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 22); // DEVPROP_TYPE_UINT32
+DEFINE_PROPERTYKEY(PKEY_Device_BusNumber, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 23); // DEVPROP_TYPE_UINT32
+DEFINE_PROPERTYKEY(PKEY_Device_EnumeratorName, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 24); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_Device_Security, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 25); // DEVPROP_TYPE_SECURITY_DESCRIPTOR
+DEFINE_PROPERTYKEY(PKEY_Device_SecuritySDS, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 26); // DEVPROP_TYPE_SECURITY_DESCRIPTOR_STRING
+DEFINE_PROPERTYKEY(PKEY_Device_DevType, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 27); // DEVPROP_TYPE_UINT32
+DEFINE_PROPERTYKEY(PKEY_Device_Exclusive, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 28); // DEVPROP_TYPE_UINT32
+DEFINE_PROPERTYKEY(PKEY_Device_Characteristics, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 29); // DEVPROP_TYPE_UINT32
+DEFINE_PROPERTYKEY(PKEY_Device_Address, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 30); // DEVPROP_TYPE_UINT32
+DEFINE_PROPERTYKEY(PKEY_Device_UINumberDescFormat, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 31); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_Device_PowerData, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 32); // DEVPROP_TYPE_BINARY
+DEFINE_PROPERTYKEY(PKEY_Device_RemovalPolicy, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 33); // DEVPROP_TYPE_UINT32
+DEFINE_PROPERTYKEY(PKEY_Device_RemovalPolicyDefault, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 34); // DEVPROP_TYPE_UINT32
+DEFINE_PROPERTYKEY(PKEY_Device_RemovalPolicyOverride, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 35); // DEVPROP_TYPE_UINT32
+DEFINE_PROPERTYKEY(PKEY_Device_InstallState, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 36); // DEVPROP_TYPE_UINT32
+DEFINE_PROPERTYKEY(PKEY_Device_LocationPaths, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 37); // DEVPROP_TYPE_STRING_LIST
+DEFINE_PROPERTYKEY(PKEY_Device_BaseContainerId, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 38); // DEVPROP_TYPE_GUID
+
+//
+// Device properties
+// These PKEYs correspond to a device's status and problem code
+//
+DEFINE_PROPERTYKEY(PKEY_Device_DevNodeStatus, 0x4340a6c5, 0x93fa, 0x4706, 0x97, 0x2c, 0x7b, 0x64, 0x80, 0x08, 0xa5, 0xa7, 2); // DEVPROP_TYPE_UINT32
+DEFINE_PROPERTYKEY(PKEY_Device_ProblemCode, 0x4340a6c5, 0x93fa, 0x4706, 0x97, 0x2c, 0x7b, 0x64, 0x80, 0x08, 0xa5, 0xa7, 3); // DEVPROP_TYPE_UINT32
+
+//
+// Device properties
+// These PKEYs correspond to device relations
+//
+DEFINE_PROPERTYKEY(PKEY_Device_EjectionRelations, 0x4340a6c5, 0x93fa, 0x4706, 0x97, 0x2c, 0x7b, 0x64, 0x80, 0x08, 0xa5, 0xa7, 4); // DEVPROP_TYPE_STRING_LIST
+DEFINE_PROPERTYKEY(PKEY_Device_RemovalRelations, 0x4340a6c5, 0x93fa, 0x4706, 0x97, 0x2c, 0x7b, 0x64, 0x80, 0x08, 0xa5, 0xa7, 5); // DEVPROP_TYPE_STRING_LIST
+DEFINE_PROPERTYKEY(PKEY_Device_PowerRelations, 0x4340a6c5, 0x93fa, 0x4706, 0x97, 0x2c, 0x7b, 0x64, 0x80, 0x08, 0xa5, 0xa7, 6); // DEVPROP_TYPE_STRING_LIST
+DEFINE_PROPERTYKEY(PKEY_Device_BusRelations, 0x4340a6c5, 0x93fa, 0x4706, 0x97, 0x2c, 0x7b, 0x64, 0x80, 0x08, 0xa5, 0xa7, 7); // DEVPROP_TYPE_STRING_LIST
+DEFINE_PROPERTYKEY(PKEY_Device_Parent, 0x4340a6c5, 0x93fa, 0x4706, 0x97, 0x2c, 0x7b, 0x64, 0x80, 0x08, 0xa5, 0xa7, 8); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_Device_Children, 0x4340a6c5, 0x93fa, 0x4706, 0x97, 0x2c, 0x7b, 0x64, 0x80, 0x08, 0xa5, 0xa7, 9); // DEVPROP_TYPE_STRING_LIST
+DEFINE_PROPERTYKEY(PKEY_Device_Siblings, 0x4340a6c5, 0x93fa, 0x4706, 0x97, 0x2c, 0x7b, 0x64, 0x80, 0x08, 0xa5, 0xa7, 10); // DEVPROP_TYPE_STRING_LIST
+DEFINE_PROPERTYKEY(PKEY_Device_TransportRelations, 0x4340a6c5, 0x93fa, 0x4706, 0x97, 0x2c, 0x7b, 0x64, 0x80, 0x08, 0xa5, 0xa7, 11); // DEVPROP_TYPE_STRING_LIST
+
+//
+// Other Device properties
+//
+DEFINE_PROPERTYKEY(PKEY_Device_Reported, 0x80497100, 0x8c73, 0x48b9, 0xaa, 0xd9, 0xce, 0x38, 0x7e, 0x19, 0xc5, 0x6e, 2); // DEVPROP_TYPE_BOOLEAN
+DEFINE_PROPERTYKEY(PKEY_Device_Legacy, 0x80497100, 0x8c73, 0x48b9, 0xaa, 0xd9, 0xce, 0x38, 0x7e, 0x19, 0xc5, 0x6e, 3); // DEVPROP_TYPE_BOOLEAN
+DEFINE_PROPERTYKEY(PKEY_Device_InstanceId, 0x78c34fc8, 0x104a, 0x4aca, 0x9e, 0xa4, 0x52, 0x4d, 0x52, 0x99, 0x6e, 0x57, 256); // DEVPROP_TYPE_STRING
+
+DEFINE_PROPERTYKEY(PKEY_Device_ContainerId, 0x8c7ed206, 0x3f8a, 0x4827, 0xb3, 0xab, 0xae, 0x9e, 0x1f, 0xae, 0xfc, 0x6c, 2); // DEVPROP_TYPE_GUID
+
+DEFINE_PROPERTYKEY(PKEY_Device_ModelId, 0x80d81ea6, 0x7473, 0x4b0c, 0x82, 0x16, 0xef, 0xc1, 0x1a, 0x2c, 0x4c, 0x8b, 2); // DEVPROP_TYPE_GUID
+
+DEFINE_PROPERTYKEY(PKEY_Device_FriendlyNameAttributes, 0x80d81ea6, 0x7473, 0x4b0c, 0x82, 0x16, 0xef, 0xc1, 0x1a, 0x2c, 0x4c, 0x8b, 3); // DEVPROP_TYPE_UINT32
+DEFINE_PROPERTYKEY(PKEY_Device_ManufacturerAttributes, 0x80d81ea6, 0x7473, 0x4b0c, 0x82, 0x16, 0xef, 0xc1, 0x1a, 0x2c, 0x4c, 0x8b, 4); // DEVPROP_TYPE_UINT32
+
+DEFINE_PROPERTYKEY(PKEY_Device_PresenceNotForDevice, 0x80d81ea6, 0x7473, 0x4b0c, 0x82, 0x16, 0xef, 0xc1, 0x1a, 0x2c, 0x4c, 0x8b, 5); // DEVPROP_TYPE_BOOLEAN
+
+
+DEFINE_PROPERTYKEY(PKEY_Numa_Proximity_Domain, 0x540b947e, 0x8b40, 0x45bc, 0xa8, 0xa2, 0x6a, 0x0b, 0x89, 0x4c, 0xbd, 0xa2, 1); // DEVPROP_TYPE_UINT32
+DEFINE_PROPERTYKEY(PKEY_Device_DHP_Rebalance_Policy, 0x540b947e, 0x8b40, 0x45bc, 0xa8, 0xa2, 0x6a, 0x0b, 0x89, 0x4c, 0xbd, 0xa2, 2); // DEVPROP_TYPE_UINT32
+DEFINE_PROPERTYKEY(PKEY_Device_Numa_Node, 0x540b947e, 0x8b40, 0x45bc, 0xa8, 0xa2, 0x6a, 0x0b, 0x89, 0x4c, 0xbd, 0xa2, 3); // DEVPROP_TYPE_UINT32
+DEFINE_PROPERTYKEY(PKEY_Device_BusReportedDeviceDesc, 0x540b947e, 0x8b40, 0x45bc, 0xa8, 0xa2, 0x6a, 0x0b, 0x89, 0x4c, 0xbd, 0xa2, 4); // DEVPROP_TYPE_STRING
+
+DEFINE_PROPERTYKEY(PKEY_Device_InstallInProgress, 0x83da6326, 0x97a6, 0x4088, 0x94, 0x53, 0xa1, 0x92, 0x3f, 0x57, 0x3b, 0x29, 9); // DEVPROP_TYPE_BOOLEAN
+
+//
+// Device driver properties
+//
+DEFINE_PROPERTYKEY(PKEY_Device_DriverDate, 0xa8b865dd, 0x2e3d, 0x4094, 0xad, 0x97, 0xe5, 0x93, 0xa7, 0xc, 0x75, 0xd6, 2); // DEVPROP_TYPE_FILETIME
+DEFINE_PROPERTYKEY(PKEY_Device_DriverVersion, 0xa8b865dd, 0x2e3d, 0x4094, 0xad, 0x97, 0xe5, 0x93, 0xa7, 0xc, 0x75, 0xd6, 3); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_Device_DriverDesc, 0xa8b865dd, 0x2e3d, 0x4094, 0xad, 0x97, 0xe5, 0x93, 0xa7, 0xc, 0x75, 0xd6, 4); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_Device_DriverInfPath, 0xa8b865dd, 0x2e3d, 0x4094, 0xad, 0x97, 0xe5, 0x93, 0xa7, 0xc, 0x75, 0xd6, 5); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_Device_DriverInfSection, 0xa8b865dd, 0x2e3d, 0x4094, 0xad, 0x97, 0xe5, 0x93, 0xa7, 0xc, 0x75, 0xd6, 6); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_Device_DriverInfSectionExt, 0xa8b865dd, 0x2e3d, 0x4094, 0xad, 0x97, 0xe5, 0x93, 0xa7, 0xc, 0x75, 0xd6, 7); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_Device_MatchingDeviceId, 0xa8b865dd, 0x2e3d, 0x4094, 0xad, 0x97, 0xe5, 0x93, 0xa7, 0xc, 0x75, 0xd6, 8); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_Device_DriverProvider, 0xa8b865dd, 0x2e3d, 0x4094, 0xad, 0x97, 0xe5, 0x93, 0xa7, 0xc, 0x75, 0xd6, 9); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_Device_DriverPropPageProvider, 0xa8b865dd, 0x2e3d, 0x4094, 0xad, 0x97, 0xe5, 0x93, 0xa7, 0xc, 0x75, 0xd6, 10); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_Device_DriverCoInstallers, 0xa8b865dd, 0x2e3d, 0x4094, 0xad, 0x97, 0xe5, 0x93, 0xa7, 0xc, 0x75, 0xd6, 11); // DEVPROP_TYPE_STRING_LIST
+DEFINE_PROPERTYKEY(PKEY_Device_ResourcePickerTags, 0xa8b865dd, 0x2e3d, 0x4094, 0xad, 0x97, 0xe5, 0x93, 0xa7, 0xc, 0x75, 0xd6, 12); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_Device_ResourcePickerExceptions, 0xa8b865dd, 0x2e3d, 0x4094, 0xad, 0x97, 0xe5, 0x93, 0xa7, 0xc, 0x75, 0xd6, 13); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_Device_DriverRank, 0xa8b865dd, 0x2e3d, 0x4094, 0xad, 0x97, 0xe5, 0x93, 0xa7, 0xc, 0x75, 0xd6, 14); // DEVPROP_TYPE_UINT32
+DEFINE_PROPERTYKEY(PKEY_Device_DriverLogoLevel, 0xa8b865dd, 0x2e3d, 0x4094, 0xad, 0x97, 0xe5, 0x93, 0xa7, 0xc, 0x75, 0xd6, 15); // DEVPROP_TYPE_UINT32
+DEFINE_PROPERTYKEY(PKEY_Device_NoConnectSound, 0xa8b865dd, 0x2e3d, 0x4094, 0xad, 0x97, 0xe5, 0x93, 0xa7, 0xc, 0x75, 0xd6, 17); // DEVPROP_TYPE_BOOLEAN
+DEFINE_PROPERTYKEY(PKEY_Device_GenericDriverInstalled, 0xa8b865dd, 0x2e3d, 0x4094, 0xad, 0x97, 0xe5, 0x93, 0xa7, 0xc, 0x75, 0xd6, 18); // DEVPROP_TYPE_BOOLEAN
+DEFINE_PROPERTYKEY(PKEY_Device_AdditionalSoftwareRequested, 0xa8b865dd, 0x2e3d, 0x4094, 0xad, 0x97, 0xe5, 0x93, 0xa7, 0xc, 0x75, 0xd6, 19);// DEVPROP_TYPE_BOOLEAN
+
+//
+// Device safe-removal properties
+//
+DEFINE_PROPERTYKEY(PKEY_Device_SafeRemovalRequired, 0xafd97640, 0x86a3, 0x4210, 0xb6, 0x7c, 0x28, 0x9c, 0x41, 0xaa, 0xbe, 0x55, 2); // DEVPROP_TYPE_BOOLEAN
+DEFINE_PROPERTYKEY(PKEY_Device_SafeRemovalRequiredOverride, 0xafd97640, 0x86a3, 0x4210, 0xb6, 0x7c, 0x28, 0x9c, 0x41, 0xaa, 0xbe, 0x55, 3);// DEVPROP_TYPE_BOOLEAN
+
+
+//
+// Device properties that were set by the driver package that was installed
+// on the device.
+//
+DEFINE_PROPERTYKEY(PKEY_DrvPkg_Model, 0xcf73bb51, 0x3abf, 0x44a2, 0x85, 0xe0, 0x9a, 0x3d, 0xc7, 0xa1, 0x21, 0x32, 2); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_DrvPkg_VendorWebSite, 0xcf73bb51, 0x3abf, 0x44a2, 0x85, 0xe0, 0x9a, 0x3d, 0xc7, 0xa1, 0x21, 0x32, 3); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_DrvPkg_DetailedDescription, 0xcf73bb51, 0x3abf, 0x44a2, 0x85, 0xe0, 0x9a, 0x3d, 0xc7, 0xa1, 0x21, 0x32, 4); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_DrvPkg_DocumentationLink, 0xcf73bb51, 0x3abf, 0x44a2, 0x85, 0xe0, 0x9a, 0x3d, 0xc7, 0xa1, 0x21, 0x32, 5); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_DrvPkg_Icon, 0xcf73bb51, 0x3abf, 0x44a2, 0x85, 0xe0, 0x9a, 0x3d, 0xc7, 0xa1, 0x21, 0x32, 6); // DEVPROP_TYPE_STRING_LIST
+DEFINE_PROPERTYKEY(PKEY_DrvPkg_BrandingIcon, 0xcf73bb51, 0x3abf, 0x44a2, 0x85, 0xe0, 0x9a, 0x3d, 0xc7, 0xa1, 0x21, 0x32, 7); // DEVPROP_TYPE_STRING_LIST
+
+//
+// Device setup class properties
+// These PKEYs correspond to the old setupapi SPCRP_XXX properties
+//
+DEFINE_PROPERTYKEY(PKEY_DeviceClass_UpperFilters, 0x4321918b, 0xf69e, 0x470d, 0xa5, 0xde, 0x4d, 0x88, 0xc7, 0x5a, 0xd2, 0x4b, 19); // DEVPROP_TYPE_STRING_LIST
+DEFINE_PROPERTYKEY(PKEY_DeviceClass_LowerFilters, 0x4321918b, 0xf69e, 0x470d, 0xa5, 0xde, 0x4d, 0x88, 0xc7, 0x5a, 0xd2, 0x4b, 20); // DEVPROP_TYPE_STRING_LIST
+DEFINE_PROPERTYKEY(PKEY_DeviceClass_Security, 0x4321918b, 0xf69e, 0x470d, 0xa5, 0xde, 0x4d, 0x88, 0xc7, 0x5a, 0xd2, 0x4b, 25); // DEVPROP_TYPE_SECURITY_DESCRIPTOR
+DEFINE_PROPERTYKEY(PKEY_DeviceClass_SecuritySDS, 0x4321918b, 0xf69e, 0x470d, 0xa5, 0xde, 0x4d, 0x88, 0xc7, 0x5a, 0xd2, 0x4b, 26); // DEVPROP_TYPE_SECURITY_DESCRIPTOR_STRING
+DEFINE_PROPERTYKEY(PKEY_DeviceClass_DevType, 0x4321918b, 0xf69e, 0x470d, 0xa5, 0xde, 0x4d, 0x88, 0xc7, 0x5a, 0xd2, 0x4b, 27); // DEVPROP_TYPE_UINT32
+DEFINE_PROPERTYKEY(PKEY_DeviceClass_Exclusive, 0x4321918b, 0xf69e, 0x470d, 0xa5, 0xde, 0x4d, 0x88, 0xc7, 0x5a, 0xd2, 0x4b, 28); // DEVPROP_TYPE_UINT32
+DEFINE_PROPERTYKEY(PKEY_DeviceClass_Characteristics, 0x4321918b, 0xf69e, 0x470d, 0xa5, 0xde, 0x4d, 0x88, 0xc7, 0x5a, 0xd2, 0x4b, 29); // DEVPROP_TYPE_UINT32
+
+//
+// Device setup class properties
+// These PKEYs correspond to registry values under the device class GUID key
+//
+DEFINE_PROPERTYKEY(PKEY_DeviceClass_Name, 0x259abffc, 0x50a7, 0x47ce, 0xaf, 0x8, 0x68, 0xc9, 0xa7, 0xd7, 0x33, 0x66, 2); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_DeviceClass_ClassName, 0x259abffc, 0x50a7, 0x47ce, 0xaf, 0x8, 0x68, 0xc9, 0xa7, 0xd7, 0x33, 0x66, 3); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_DeviceClass_Icon, 0x259abffc, 0x50a7, 0x47ce, 0xaf, 0x8, 0x68, 0xc9, 0xa7, 0xd7, 0x33, 0x66, 4); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_DeviceClass_ClassInstaller, 0x259abffc, 0x50a7, 0x47ce, 0xaf, 0x8, 0x68, 0xc9, 0xa7, 0xd7, 0x33, 0x66, 5); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_DeviceClass_PropPageProvider, 0x259abffc, 0x50a7, 0x47ce, 0xaf, 0x8, 0x68, 0xc9, 0xa7, 0xd7, 0x33, 0x66, 6); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_DeviceClass_NoInstallClass, 0x259abffc, 0x50a7, 0x47ce, 0xaf, 0x8, 0x68, 0xc9, 0xa7, 0xd7, 0x33, 0x66, 7); // DEVPROP_TYPE_BOOLEAN
+DEFINE_PROPERTYKEY(PKEY_DeviceClass_NoDisplayClass, 0x259abffc, 0x50a7, 0x47ce, 0xaf, 0x8, 0x68, 0xc9, 0xa7, 0xd7, 0x33, 0x66, 8); // DEVPROP_TYPE_BOOLEAN
+DEFINE_PROPERTYKEY(PKEY_DeviceClass_SilentInstall, 0x259abffc, 0x50a7, 0x47ce, 0xaf, 0x8, 0x68, 0xc9, 0xa7, 0xd7, 0x33, 0x66, 9); // DEVPROP_TYPE_BOOLEAN
+DEFINE_PROPERTYKEY(PKEY_DeviceClass_NoUseClass, 0x259abffc, 0x50a7, 0x47ce, 0xaf, 0x8, 0x68, 0xc9, 0xa7, 0xd7, 0x33, 0x66, 10); // DEVPROP_TYPE_BOOLEAN
+DEFINE_PROPERTYKEY(PKEY_DeviceClass_DefaultService, 0x259abffc, 0x50a7, 0x47ce, 0xaf, 0x8, 0x68, 0xc9, 0xa7, 0xd7, 0x33, 0x66, 11); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_DeviceClass_IconPath, 0x259abffc, 0x50a7, 0x47ce, 0xaf, 0x8, 0x68, 0xc9, 0xa7, 0xd7, 0x33, 0x66, 12); // DEVPROP_TYPE_STRING_LIST
+
+//
+// Other Device setup class properties
+//
+DEFINE_PROPERTYKEY(PKEY_DeviceClass_ClassCoInstallers, 0x713d1703, 0xa2e2, 0x49f5, 0x92, 0x14, 0x56, 0x47, 0x2e, 0xf3, 0xda, 0x5c, 2); // DEVPROP_TYPE_STRING_LIST
+
+//
+// Device interface properties
+//
+DEFINE_PROPERTYKEY(PKEY_DeviceInterface_FriendlyName, 0x026e516e, 0xb814, 0x414b, 0x83, 0xcd, 0x85, 0x6d, 0x6f, 0xef, 0x48, 0x22, 2); // DEVPROP_TYPE_STRING
+DEFINE_PROPERTYKEY(PKEY_DeviceInterface_Enabled, 0x026e516e, 0xb814, 0x414b, 0x83, 0xcd, 0x85, 0x6d, 0x6f, 0xef, 0x48, 0x22, 3); // DEVPROP_TYPE_BOOLEAN
+DEFINE_PROPERTYKEY(PKEY_DeviceInterface_ClassGuid, 0x026e516e, 0xb814, 0x414b, 0x83, 0xcd, 0x85, 0x6d, 0x6f, 0xef, 0x48, 0x22, 4); // DEVPROP_TYPE_GUID
+
+//
+// Device interface class properties
+//
+DEFINE_PROPERTYKEY(PKEY_DeviceInterfaceClass_DefaultInterface, 0x14c83a99, 0x0b3f, 0x44b7, 0xbe, 0x4c, 0xa1, 0x78, 0xd3, 0x99, 0x05, 0x64, 2); // DEVPROP_TYPE_STRING
+
+
+
+
diff --git a/RtAudio/RtAudio.cpp b/RtAudio/RtAudio.cpp
new file mode 100644
index 0000000..af61bc7
--- /dev/null
+++ b/RtAudio/RtAudio.cpp
@@ -0,0 +1,10229 @@
+/************************************************************************/
+/*! \class RtAudio
+ \brief Realtime audio i/o C++ classes.
+
+ RtAudio provides a common API (Application Programming Interface)
+ for realtime audio input/output across Linux (native ALSA, Jack,
+ and OSS), Macintosh OS X (CoreAudio and Jack), and Windows
+ (DirectSound, ASIO and WASAPI) operating systems.
+
+ RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
+
+ RtAudio: realtime audio i/o C++ classes
+ Copyright (c) 2001-2016 Gary P. Scavone
+
+ Permission is hereby granted, free of charge, to any person
+ obtaining a copy of this software and associated documentation files
+ (the "Software"), to deal in the Software without restriction,
+ including without limitation the rights to use, copy, modify, merge,
+ publish, distribute, sublicense, and/or sell copies of the Software,
+ and to permit persons to whom the Software is furnished to do so,
+ subject to the following conditions:
+
+ The above copyright notice and this permission notice shall be
+ included in all copies or substantial portions of the Software.
+
+ Any person wishing to distribute modifications to the Software is
+ asked to send the modifications to the original developer so that
+ they can be incorporated into the canonical version. This is,
+ however, not a binding provision of this license.
+
+ THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+ EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+ ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+ CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+ WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+*/
+/************************************************************************/
+
+// RtAudio: Version 4.1.2
+
+#include "RtAudio.h"
+#include <iostream>
+#include <cstdlib>
+#include <cstring>
+#include <climits>
+#include <algorithm>
+
+// Static variable definitions.
+const unsigned int RtApi::MAX_SAMPLE_RATES = 14;
+const unsigned int RtApi::SAMPLE_RATES[] = {
+ 4000, 5512, 8000, 9600, 11025, 16000, 22050,
+ 32000, 44100, 48000, 88200, 96000, 176400, 192000
+};
+
+#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__)
+ #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A)
+ #define MUTEX_DESTROY(A) DeleteCriticalSection(A)
+ #define MUTEX_LOCK(A) EnterCriticalSection(A)
+ #define MUTEX_UNLOCK(A) LeaveCriticalSection(A)
+
+ #include "tchar.h"
+
+ static std::string convertCharPointerToStdString(const char *text)
+ {
+ return std::string(text);
+ }
+
+ static std::string convertCharPointerToStdString(const wchar_t *text)
+ {
+ int length = WideCharToMultiByte(CP_UTF8, 0, text, -1, NULL, 0, NULL, NULL);
+ std::string s( length-1, '\0' );
+ WideCharToMultiByte(CP_UTF8, 0, text, -1, &s[0], length, NULL, NULL);
+ return s;
+ }
+
+#elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
+ // pthread API
+ #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL)
+ #define MUTEX_DESTROY(A) pthread_mutex_destroy(A)
+ #define MUTEX_LOCK(A) pthread_mutex_lock(A)
+ #define MUTEX_UNLOCK(A) pthread_mutex_unlock(A)
+#else
+ #define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions
+ #define MUTEX_DESTROY(A) abs(*A) // dummy definitions
+#endif
+
+// *************************************************** //
+//
+// RtAudio definitions.
+//
+// *************************************************** //
+
+std::string RtAudio :: getVersion( void ) throw()
+{
+ return RTAUDIO_VERSION;
+}
+
+void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis ) throw()
+{
+ apis.clear();
+
+ // The order here will control the order of RtAudio's API search in
+ // the constructor.
+#if defined(__UNIX_JACK__)
+ apis.push_back( UNIX_JACK );
+#endif
+#if defined(__LINUX_ALSA__)
+ apis.push_back( LINUX_ALSA );
+#endif
+#if defined(__LINUX_PULSE__)
+ apis.push_back( LINUX_PULSE );
+#endif
+#if defined(__LINUX_OSS__)
+ apis.push_back( LINUX_OSS );
+#endif
+#if defined(__WINDOWS_ASIO__)
+ apis.push_back( WINDOWS_ASIO );
+#endif
+#if defined(__WINDOWS_WASAPI__)
+ apis.push_back( WINDOWS_WASAPI );
+#endif
+#if defined(__WINDOWS_DS__)
+ apis.push_back( WINDOWS_DS );
+#endif
+#if defined(__MACOSX_CORE__)
+ apis.push_back( MACOSX_CORE );
+#endif
+#if defined(__RTAUDIO_DUMMY__)
+ apis.push_back( RTAUDIO_DUMMY );
+#endif
+}
+
+void RtAudio :: openRtApi( RtAudio::Api api )
+{
+ if ( rtapi_ )
+ delete rtapi_;
+ rtapi_ = 0;
+
+#if defined(__UNIX_JACK__)
+ if ( api == UNIX_JACK )
+ rtapi_ = new RtApiJack();
+#endif
+#if defined(__LINUX_ALSA__)
+ if ( api == LINUX_ALSA )
+ rtapi_ = new RtApiAlsa();
+#endif
+#if defined(__LINUX_PULSE__)
+ if ( api == LINUX_PULSE )
+ rtapi_ = new RtApiPulse();
+#endif
+#if defined(__LINUX_OSS__)
+ if ( api == LINUX_OSS )
+ rtapi_ = new RtApiOss();
+#endif
+#if defined(__WINDOWS_ASIO__)
+ if ( api == WINDOWS_ASIO )
+ rtapi_ = new RtApiAsio();
+#endif
+#if defined(__WINDOWS_WASAPI__)
+ if ( api == WINDOWS_WASAPI )
+ rtapi_ = new RtApiWasapi();
+#endif
+#if defined(__WINDOWS_DS__)
+ if ( api == WINDOWS_DS )
+ rtapi_ = new RtApiDs();
+#endif
+#if defined(__MACOSX_CORE__)
+ if ( api == MACOSX_CORE )
+ rtapi_ = new RtApiCore();
+#endif
+#if defined(__RTAUDIO_DUMMY__)
+ if ( api == RTAUDIO_DUMMY )
+ rtapi_ = new RtApiDummy();
+#endif
+}
+
+RtAudio :: RtAudio( RtAudio::Api api )
+{
+ rtapi_ = 0;
+
+ if ( api != UNSPECIFIED ) {
+ // Attempt to open the specified API.
+ openRtApi( api );
+ if ( rtapi_ ) return;
+
+ // No compiled support for specified API value. Issue a debug
+ // warning and continue as if no API was specified.
+ std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl;
+ }
+
+ // Iterate through the compiled APIs and return as soon as we find
+ // one with at least one device or we reach the end of the list.
+ std::vector< RtAudio::Api > apis;
+ getCompiledApi( apis );
+ for ( unsigned int i=0; i<apis.size(); i++ ) {
+ openRtApi( apis[i] );
+ if ( rtapi_ && rtapi_->getDeviceCount() ) break;
+ }
+
+ if ( rtapi_ ) return;
+
+ // It should not be possible to get here because the preprocessor
+ // definition __RTAUDIO_DUMMY__ is automatically defined if no
+ // API-specific definitions are passed to the compiler. But just in
+ // case something weird happens, we'll thow an error.
+ std::string errorText = "\nRtAudio: no compiled API support found ... critical error!!\n\n";
+ throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) );
+}
+
+RtAudio :: ~RtAudio() throw()
+{
+ if ( rtapi_ )
+ delete rtapi_;
+}
+
+void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters,
+ RtAudio::StreamParameters *inputParameters,
+ RtAudioFormat format, unsigned int sampleRate,
+ unsigned int *bufferFrames,
+ RtAudioCallback callback, void *userData,
+ RtAudio::StreamOptions *options,
+ RtAudioErrorCallback errorCallback )
+{
+ return rtapi_->openStream( outputParameters, inputParameters, format,
+ sampleRate, bufferFrames, callback,
+ userData, options, errorCallback );
+}
+
+// *************************************************** //
+//
+// Public RtApi definitions (see end of file for
+// private or protected utility functions).
+//
+// *************************************************** //
+
+RtApi :: RtApi()
+{
+ stream_.state = STREAM_CLOSED;
+ stream_.mode = UNINITIALIZED;
+ stream_.apiHandle = 0;
+ stream_.userBuffer[0] = 0;
+ stream_.userBuffer[1] = 0;
+ MUTEX_INITIALIZE( &stream_.mutex );
+ showWarnings_ = true;
+ firstErrorOccurred_ = false;
+}
+
+RtApi :: ~RtApi()
+{
+ MUTEX_DESTROY( &stream_.mutex );
+}
+
+void RtApi :: openStream( RtAudio::StreamParameters *oParams,
+ RtAudio::StreamParameters *iParams,
+ RtAudioFormat format, unsigned int sampleRate,
+ unsigned int *bufferFrames,
+ RtAudioCallback callback, void *userData,
+ RtAudio::StreamOptions *options,
+ RtAudioErrorCallback errorCallback )
+{
+ if ( stream_.state != STREAM_CLOSED ) {
+ errorText_ = "RtApi::openStream: a stream is already open!";
+ error( RtAudioError::INVALID_USE );
+ return;
+ }
+
+ // Clear stream information potentially left from a previously open stream.
+ clearStreamInfo();
+
+ if ( oParams && oParams->nChannels < 1 ) {
+ errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one.";
+ error( RtAudioError::INVALID_USE );
+ return;
+ }
+
+ if ( iParams && iParams->nChannels < 1 ) {
+ errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one.";
+ error( RtAudioError::INVALID_USE );
+ return;
+ }
+
+ if ( oParams == NULL && iParams == NULL ) {
+ errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!";
+ error( RtAudioError::INVALID_USE );
+ return;
+ }
+
+ if ( formatBytes(format) == 0 ) {
+ errorText_ = "RtApi::openStream: 'format' parameter value is undefined.";
+ error( RtAudioError::INVALID_USE );
+ return;
+ }
+
+ unsigned int nDevices = getDeviceCount();
+ unsigned int oChannels = 0;
+ if ( oParams ) {
+ oChannels = oParams->nChannels;
+ if ( oParams->deviceId >= nDevices ) {
+ errorText_ = "RtApi::openStream: output device parameter value is invalid.";
+ error( RtAudioError::INVALID_USE );
+ return;
+ }
+ }
+
+ unsigned int iChannels = 0;
+ if ( iParams ) {
+ iChannels = iParams->nChannels;
+ if ( iParams->deviceId >= nDevices ) {
+ errorText_ = "RtApi::openStream: input device parameter value is invalid.";
+ error( RtAudioError::INVALID_USE );
+ return;
+ }
+ }
+
+ bool result;
+
+ if ( oChannels > 0 ) {
+
+ result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel,
+ sampleRate, format, bufferFrames, options );
+ if ( result == false ) {
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ }
+
+ if ( iChannels > 0 ) {
+
+ result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel,
+ sampleRate, format, bufferFrames, options );
+ if ( result == false ) {
+ if ( oChannels > 0 ) closeStream();
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ }
+
+ stream_.callbackInfo.callback = (void *) callback;
+ stream_.callbackInfo.userData = userData;
+ stream_.callbackInfo.errorCallback = (void *) errorCallback;
+
+ if ( options ) options->numberOfBuffers = stream_.nBuffers;
+ stream_.state = STREAM_STOPPED;
+}
+
+unsigned int RtApi :: getDefaultInputDevice( void )
+{
+ // Should be implemented in subclasses if possible.
+ return 0;
+}
+
+unsigned int RtApi :: getDefaultOutputDevice( void )
+{
+ // Should be implemented in subclasses if possible.
+ return 0;
+}
+
+void RtApi :: closeStream( void )
+{
+ // MUST be implemented in subclasses!
+ return;
+}
+
+bool RtApi :: probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/,
+ unsigned int /*firstChannel*/, unsigned int /*sampleRate*/,
+ RtAudioFormat /*format*/, unsigned int * /*bufferSize*/,
+ RtAudio::StreamOptions * /*options*/ )
+{
+ // MUST be implemented in subclasses!
+ return FAILURE;
+}
+
+void RtApi :: tickStreamTime( void )
+{
+ // Subclasses that do not provide their own implementation of
+ // getStreamTime should call this function once per buffer I/O to
+ // provide basic stream time support.
+
+ stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate );
+
+#if defined( HAVE_GETTIMEOFDAY )
+ gettimeofday( &stream_.lastTickTimestamp, NULL );
+#endif
+}
+
+long RtApi :: getStreamLatency( void )
+{
+ verifyStream();
+
+ long totalLatency = 0;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
+ totalLatency = stream_.latency[0];
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
+ totalLatency += stream_.latency[1];
+
+ return totalLatency;
+}
+
+double RtApi :: getStreamTime( void )
+{
+ verifyStream();
+
+#if defined( HAVE_GETTIMEOFDAY )
+ // Return a very accurate estimate of the stream time by
+ // adding in the elapsed time since the last tick.
+ struct timeval then;
+ struct timeval now;
+
+ if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 )
+ return stream_.streamTime;
+
+ gettimeofday( &now, NULL );
+ then = stream_.lastTickTimestamp;
+ return stream_.streamTime +
+ ((now.tv_sec + 0.000001 * now.tv_usec) -
+ (then.tv_sec + 0.000001 * then.tv_usec));
+#else
+ return stream_.streamTime;
+#endif
+}
+
+void RtApi :: setStreamTime( double time )
+{
+ verifyStream();
+
+ if ( time >= 0.0 )
+ stream_.streamTime = time;
+}
+
+unsigned int RtApi :: getStreamSampleRate( void )
+{
+ verifyStream();
+
+ return stream_.sampleRate;
+}
+
+
+// *************************************************** //
+//
+// OS/API-specific methods.
+//
+// *************************************************** //
+
+#if defined(__MACOSX_CORE__)
+
+// The OS X CoreAudio API is designed to use a separate callback
+// procedure for each of its audio devices. A single RtAudio duplex
+// stream using two different devices is supported here, though it
+// cannot be guaranteed to always behave correctly because we cannot
+// synchronize these two callbacks.
+//
+// A property listener is installed for over/underrun information.
+// However, no functionality is currently provided to allow property
+// listeners to trigger user handlers because it is unclear what could
+// be done if a critical stream parameter (buffer size, sample rate,
+// device disconnect) notification arrived. The listeners entail
+// quite a bit of extra code and most likely, a user program wouldn't
+// be prepared for the result anyway. However, we do provide a flag
+// to the client callback function to inform of an over/underrun.
+
+// A structure to hold various information related to the CoreAudio API
+// implementation.
+struct CoreHandle {
+ AudioDeviceID id[2]; // device ids
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+ AudioDeviceIOProcID procId[2];
+#endif
+ UInt32 iStream[2]; // device stream index (or first if using multiple)
+ UInt32 nStreams[2]; // number of streams to use
+ bool xrun[2];
+ char *deviceBuffer;
+ pthread_cond_t condition;
+ int drainCounter; // Tracks callback counts when draining
+ bool internalDrain; // Indicates if stop is initiated from callback or not.
+
+ CoreHandle()
+ :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
+};
+
+RtApiCore:: RtApiCore()
+{
+#if defined( AVAILABLE_MAC_OS_X_VERSION_10_6_AND_LATER )
+ // This is a largely undocumented but absolutely necessary
+ // requirement starting with OS-X 10.6. If not called, queries and
+ // updates to various audio device properties are not handled
+ // correctly.
+ CFRunLoopRef theRunLoop = NULL;
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyRunLoop,
+ kAudioObjectPropertyScopeGlobal,
+ kAudioObjectPropertyElementMaster };
+ OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop);
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::RtApiCore: error setting run loop property!";
+ error( RtAudioError::WARNING );
+ }
+#endif
+}
+
+RtApiCore :: ~RtApiCore()
+{
+ // The subclass destructor gets called before the base class
+ // destructor, so close an existing stream before deallocating
+ // apiDeviceId memory.
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
+}
+
+unsigned int RtApiCore :: getDeviceCount( void )
+{
+ // Find out how many audio devices there are, if any.
+ UInt32 dataSize;
+ AudioObjectPropertyAddress propertyAddress = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
+ OSStatus result = AudioObjectGetPropertyDataSize( kAudioObjectSystemObject, &propertyAddress, 0, NULL, &dataSize );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!";
+ error( RtAudioError::WARNING );
+ return 0;
+ }
+
+ return dataSize / sizeof( AudioDeviceID );
+}
+
+unsigned int RtApiCore :: getDefaultInputDevice( void )
+{
+ unsigned int nDevices = getDeviceCount();
+ if ( nDevices <= 1 ) return 0;
+
+ AudioDeviceID id;
+ UInt32 dataSize = sizeof( AudioDeviceID );
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
+ OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device.";
+ error( RtAudioError::WARNING );
+ return 0;
+ }
+
+ dataSize *= nDevices;
+ AudioDeviceID deviceList[ nDevices ];
+ property.mSelector = kAudioHardwarePropertyDevices;
+ result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs.";
+ error( RtAudioError::WARNING );
+ return 0;
+ }
+
+ for ( unsigned int i=0; i<nDevices; i++ )
+ if ( id == deviceList[i] ) return i;
+
+ errorText_ = "RtApiCore::getDefaultInputDevice: No default device found!";
+ error( RtAudioError::WARNING );
+ return 0;
+}
+
+unsigned int RtApiCore :: getDefaultOutputDevice( void )
+{
+ unsigned int nDevices = getDeviceCount();
+ if ( nDevices <= 1 ) return 0;
+
+ AudioDeviceID id;
+ UInt32 dataSize = sizeof( AudioDeviceID );
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultOutputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
+ OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device.";
+ error( RtAudioError::WARNING );
+ return 0;
+ }
+
+ dataSize = sizeof( AudioDeviceID ) * nDevices;
+ AudioDeviceID deviceList[ nDevices ];
+ property.mSelector = kAudioHardwarePropertyDevices;
+ result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device IDs.";
+ error( RtAudioError::WARNING );
+ return 0;
+ }
+
+ for ( unsigned int i=0; i<nDevices; i++ )
+ if ( id == deviceList[i] ) return i;
+
+ errorText_ = "RtApiCore::getDefaultOutputDevice: No default device found!";
+ error( RtAudioError::WARNING );
+ return 0;
+}
+
+RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device )
+{
+ RtAudio::DeviceInfo info;
+ info.probed = false;
+
+ // Get device ID
+ unsigned int nDevices = getDeviceCount();
+ if ( nDevices == 0 ) {
+ errorText_ = "RtApiCore::getDeviceInfo: no devices found!";
+ error( RtAudioError::INVALID_USE );
+ return info;
+ }
+
+ if ( device >= nDevices ) {
+ errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!";
+ error( RtAudioError::INVALID_USE );
+ return info;
+ }
+
+ AudioDeviceID deviceList[ nDevices ];
+ UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
+ kAudioObjectPropertyScopeGlobal,
+ kAudioObjectPropertyElementMaster };
+ OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
+ 0, NULL, &dataSize, (void *) &deviceList );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs.";
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ AudioDeviceID id = deviceList[ device ];
+
+ // Get the device name.
+ info.name.erase();
+ CFStringRef cfname;
+ dataSize = sizeof( CFStringRef );
+ property.mSelector = kAudioObjectPropertyManufacturer;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer.";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ //const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
+ int length = CFStringGetLength(cfname);
+ char *mname = (char *)malloc(length * 3 + 1);
+#if defined( UNICODE ) || defined( _UNICODE )
+ CFStringGetCString(cfname, mname, length * 3 + 1, kCFStringEncodingUTF8);
+#else
+ CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding());
+#endif
+ info.name.append( (const char *)mname, strlen(mname) );
+ info.name.append( ": " );
+ CFRelease( cfname );
+ free(mname);
+
+ property.mSelector = kAudioObjectPropertyName;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name.";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ //const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
+ length = CFStringGetLength(cfname);
+ char *name = (char *)malloc(length * 3 + 1);
+#if defined( UNICODE ) || defined( _UNICODE )
+ CFStringGetCString(cfname, name, length * 3 + 1, kCFStringEncodingUTF8);
+#else
+ CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding());
+#endif
+ info.name.append( (const char *)name, strlen(name) );
+ CFRelease( cfname );
+ free(name);
+
+ // Get the output stream "configuration".
+ AudioBufferList *bufferList = nil;
+ property.mSelector = kAudioDevicePropertyStreamConfiguration;
+ property.mScope = kAudioDevicePropertyScopeOutput;
+ // property.mElement = kAudioObjectPropertyElementWildcard;
+ dataSize = 0;
+ result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
+ if ( result != noErr || dataSize == 0 ) {
+ errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // Allocate the AudioBufferList.
+ bufferList = (AudioBufferList *) malloc( dataSize );
+ if ( bufferList == NULL ) {
+ errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList.";
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
+ if ( result != noErr || dataSize == 0 ) {
+ free( bufferList );
+ errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // Get output channel information.
+ unsigned int i, nStreams = bufferList->mNumberBuffers;
+ for ( i=0; i<nStreams; i++ )
+ info.outputChannels += bufferList->mBuffers[i].mNumberChannels;
+ free( bufferList );
+
+ // Get the input stream "configuration".
+ property.mScope = kAudioDevicePropertyScopeInput;
+ result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
+ if ( result != noErr || dataSize == 0 ) {
+ errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // Allocate the AudioBufferList.
+ bufferList = (AudioBufferList *) malloc( dataSize );
+ if ( bufferList == NULL ) {
+ errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList.";
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
+ if (result != noErr || dataSize == 0) {
+ free( bufferList );
+ errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // Get input channel information.
+ nStreams = bufferList->mNumberBuffers;
+ for ( i=0; i<nStreams; i++ )
+ info.inputChannels += bufferList->mBuffers[i].mNumberChannels;
+ free( bufferList );
+
+ // 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;
+
+ // Probe the device sample rates.
+ bool isInput = false;
+ if ( info.outputChannels == 0 ) isInput = true;
+
+ // Determine the supported sample rates.
+ property.mSelector = kAudioDevicePropertyAvailableNominalSampleRates;
+ if ( isInput == false ) property.mScope = kAudioDevicePropertyScopeOutput;
+ result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
+ if ( result != kAudioHardwareNoError || dataSize == 0 ) {
+ errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info.";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ UInt32 nRanges = dataSize / sizeof( AudioValueRange );
+ AudioValueRange rangeList[ nRanges ];
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &rangeList );
+ if ( result != kAudioHardwareNoError ) {
+ errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates.";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // The sample rate reporting mechanism is a bit of a mystery. It
+ // seems that it can either return individual rates or a range of
+ // rates. I assume that if the min / max range values are the same,
+ // then that represents a single supported rate and if the min / max
+ // range values are different, the device supports an arbitrary
+ // range of values (though there might be multiple ranges, so we'll
+ // use the most conservative range).
+ Float64 minimumRate = 1.0, maximumRate = 10000000000.0;
+ bool haveValueRange = false;
+ info.sampleRates.clear();
+ for ( UInt32 i=0; i<nRanges; i++ ) {
+ if ( rangeList[i].mMinimum == rangeList[i].mMaximum ) {
+ unsigned int tmpSr = (unsigned int) rangeList[i].mMinimum;
+ info.sampleRates.push_back( tmpSr );
+
+ if ( !info.preferredSampleRate || ( tmpSr <= 48000 && tmpSr > info.preferredSampleRate ) )
+ info.preferredSampleRate = tmpSr;
+
+ } else {
+ haveValueRange = true;
+ if ( rangeList[i].mMinimum > minimumRate ) minimumRate = rangeList[i].mMinimum;
+ if ( rangeList[i].mMaximum < maximumRate ) maximumRate = rangeList[i].mMaximum;
+ }
+ }
+
+ if ( haveValueRange ) {
+ for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
+ if ( SAMPLE_RATES[k] >= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate ) {
+ info.sampleRates.push_back( SAMPLE_RATES[k] );
+
+ if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
+ info.preferredSampleRate = SAMPLE_RATES[k];
+ }
+ }
+ }
+
+ // Sort and remove any redundant values
+ std::sort( info.sampleRates.begin(), info.sampleRates.end() );
+ info.sampleRates.erase( unique( info.sampleRates.begin(), info.sampleRates.end() ), info.sampleRates.end() );
+
+ if ( info.sampleRates.size() == 0 ) {
+ errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // CoreAudio always uses 32-bit floating point data for PCM streams.
+ // Thus, any other "physical" formats supported by the device are of
+ // no interest to the client.
+ info.nativeFormats = RTAUDIO_FLOAT32;
+
+ if ( info.outputChannels > 0 )
+ if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
+ if ( info.inputChannels > 0 )
+ if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
+
+ info.probed = true;
+ return info;
+}
+
+static OSStatus callbackHandler( AudioDeviceID inDevice,
+ const AudioTimeStamp* /*inNow*/,
+ const AudioBufferList* inInputData,
+ const AudioTimeStamp* /*inInputTime*/,
+ AudioBufferList* outOutputData,
+ const AudioTimeStamp* /*inOutputTime*/,
+ void* infoPointer )
+{
+ CallbackInfo *info = (CallbackInfo *) infoPointer;
+
+ RtApiCore *object = (RtApiCore *) info->object;
+ if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false )
+ return kAudioHardwareUnspecifiedError;
+ else
+ return kAudioHardwareNoError;
+}
+
+static OSStatus xrunListener( AudioObjectID /*inDevice*/,
+ UInt32 nAddresses,
+ const AudioObjectPropertyAddress properties[],
+ void* handlePointer )
+{
+ CoreHandle *handle = (CoreHandle *) handlePointer;
+ for ( UInt32 i=0; i<nAddresses; i++ ) {
+ if ( properties[i].mSelector == kAudioDeviceProcessorOverload ) {
+ if ( properties[i].mScope == kAudioDevicePropertyScopeInput )
+ handle->xrun[1] = true;
+ else
+ handle->xrun[0] = true;
+ }
+ }
+
+ return kAudioHardwareNoError;
+}
+
+static OSStatus rateListener( AudioObjectID inDevice,
+ UInt32 /*nAddresses*/,
+ const AudioObjectPropertyAddress /*properties*/[],
+ void* ratePointer )
+{
+ Float64 *rate = (Float64 *) ratePointer;
+ UInt32 dataSize = sizeof( Float64 );
+ AudioObjectPropertyAddress property = { kAudioDevicePropertyNominalSampleRate,
+ kAudioObjectPropertyScopeGlobal,
+ kAudioObjectPropertyElementMaster };
+ AudioObjectGetPropertyData( inDevice, &property, 0, NULL, &dataSize, rate );
+ return kAudioHardwareNoError;
+}
+
+bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options )
+{
+ // Get device ID
+ unsigned int nDevices = getDeviceCount();
+ if ( nDevices == 0 ) {
+ // This should not happen because a check is made before this function is called.
+ errorText_ = "RtApiCore::probeDeviceOpen: no devices found!";
+ return FAILURE;
+ }
+
+ if ( device >= nDevices ) {
+ // This should not happen because a check is made before this function is called.
+ errorText_ = "RtApiCore::probeDeviceOpen: device ID is invalid!";
+ return FAILURE;
+ }
+
+ AudioDeviceID deviceList[ nDevices ];
+ UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
+ kAudioObjectPropertyScopeGlobal,
+ kAudioObjectPropertyElementMaster };
+ OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
+ 0, NULL, &dataSize, (void *) &deviceList );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs.";
+ return FAILURE;
+ }
+
+ AudioDeviceID id = deviceList[ device ];
+
+ // Setup for stream mode.
+ bool isInput = false;
+ if ( mode == INPUT ) {
+ isInput = true;
+ property.mScope = kAudioDevicePropertyScopeInput;
+ }
+ else
+ property.mScope = kAudioDevicePropertyScopeOutput;
+
+ // Get the stream "configuration".
+ AudioBufferList *bufferList = nil;
+ dataSize = 0;
+ property.mSelector = kAudioDevicePropertyStreamConfiguration;
+ result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
+ if ( result != noErr || dataSize == 0 ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration info for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Allocate the AudioBufferList.
+ bufferList = (AudioBufferList *) malloc( dataSize );
+ if ( bufferList == NULL ) {
+ errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList.";
+ return FAILURE;
+ }
+
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
+ if (result != noErr || dataSize == 0) {
+ free( bufferList );
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Search for one or more streams that contain the desired number of
+ // channels. CoreAudio devices can have an arbitrary number of
+ // streams and each stream can have an arbitrary number of channels.
+ // For each stream, a single buffer of interleaved samples is
+ // provided. RtAudio prefers the use of one stream of interleaved
+ // data or multiple consecutive single-channel streams. However, we
+ // now support multiple consecutive multi-channel streams of
+ // interleaved data as well.
+ UInt32 iStream, offsetCounter = firstChannel;
+ UInt32 nStreams = bufferList->mNumberBuffers;
+ bool monoMode = false;
+ bool foundStream = false;
+
+ // First check that the device supports the requested number of
+ // channels.
+ UInt32 deviceChannels = 0;
+ for ( iStream=0; iStream<nStreams; iStream++ )
+ deviceChannels += bufferList->mBuffers[iStream].mNumberChannels;
+
+ if ( deviceChannels < ( channels + firstChannel ) ) {
+ free( bufferList );
+ errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Look for a single stream meeting our needs.
+ UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0;
+ for ( iStream=0; iStream<nStreams; iStream++ ) {
+ streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
+ if ( streamChannels >= channels + offsetCounter ) {
+ firstStream = iStream;
+ channelOffset = offsetCounter;
+ foundStream = true;
+ break;
+ }
+ if ( streamChannels > offsetCounter ) break;
+ offsetCounter -= streamChannels;
+ }
+
+ // If we didn't find a single stream above, then we should be able
+ // to meet the channel specification with multiple streams.
+ if ( foundStream == false ) {
+ monoMode = true;
+ offsetCounter = firstChannel;
+ for ( iStream=0; iStream<nStreams; iStream++ ) {
+ streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
+ if ( streamChannels > offsetCounter ) break;
+ offsetCounter -= streamChannels;
+ }
+
+ firstStream = iStream;
+ channelOffset = offsetCounter;
+ Int32 channelCounter = channels + offsetCounter - streamChannels;
+
+ if ( streamChannels > 1 ) monoMode = false;
+ while ( channelCounter > 0 ) {
+ streamChannels = bufferList->mBuffers[++iStream].mNumberChannels;
+ if ( streamChannels > 1 ) monoMode = false;
+ channelCounter -= streamChannels;
+ streamCount++;
+ }
+ }
+
+ free( bufferList );
+
+ // Determine the buffer size.
+ AudioValueRange bufferRange;
+ dataSize = sizeof( AudioValueRange );
+ property.mSelector = kAudioDevicePropertyBufferFrameSizeRange;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &bufferRange );
+
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum;
+ else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum;
+ if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum;
+
+ // Set the buffer size. For multiple streams, I'm assuming we only
+ // need to make this setting for the master channel.
+ UInt32 theSize = (UInt32) *bufferSize;
+ dataSize = sizeof( UInt32 );
+ property.mSelector = kAudioDevicePropertyBufferFrameSize;
+ result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &theSize );
+
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // If attempting to setup a duplex stream, the bufferSize parameter
+ // MUST be the same in both directions!
+ *bufferSize = theSize;
+ if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ stream_.bufferSize = *bufferSize;
+ stream_.nBuffers = 1;
+
+ // Try to set "hog" mode ... it's not clear to me this is working.
+ if ( options && options->flags & RTAUDIO_HOG_DEVICE ) {
+ pid_t hog_pid;
+ dataSize = sizeof( hog_pid );
+ property.mSelector = kAudioDevicePropertyHogMode;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &hog_pid );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting 'hog' state!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ if ( hog_pid != getpid() ) {
+ hog_pid = getpid();
+ result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &hog_pid );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+ }
+
+ // Check and if necessary, change the sample rate for the device.
+ Float64 nominalRate;
+ dataSize = sizeof( Float64 );
+ property.mSelector = kAudioDevicePropertyNominalSampleRate;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &nominalRate );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting current sample rate.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Only change the sample rate if off by more than 1 Hz.
+ if ( fabs( nominalRate - (double)sampleRate ) > 1.0 ) {
+
+ // Set a property listener for the sample rate change
+ Float64 reportedRate = 0.0;
+ AudioObjectPropertyAddress tmp = { kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
+ result = AudioObjectAddPropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate property listener for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ nominalRate = (Float64) sampleRate;
+ result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate );
+ if ( result != noErr ) {
+ AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Now wait until the reported nominal rate is what we just set.
+ UInt32 microCounter = 0;
+ while ( reportedRate != nominalRate ) {
+ microCounter += 5000;
+ if ( microCounter > 5000000 ) break;
+ usleep( 5000 );
+ }
+
+ // Remove the property listener.
+ AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
+
+ if ( microCounter > 5000000 ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: timeout waiting for sample rate update for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+
+ // Now set the stream format for all streams. Also, check the
+ // physical format of the device and change that if necessary.
+ AudioStreamBasicDescription description;
+ dataSize = sizeof( AudioStreamBasicDescription );
+ property.mSelector = kAudioStreamPropertyVirtualFormat;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Set the sample rate and data format id. However, only make the
+ // change if the sample rate is not within 1.0 of the desired
+ // rate and the format is not linear pcm.
+ bool updateFormat = false;
+ if ( fabs( description.mSampleRate - (Float64)sampleRate ) > 1.0 ) {
+ description.mSampleRate = (Float64) sampleRate;
+ updateFormat = true;
+ }
+
+ if ( description.mFormatID != kAudioFormatLinearPCM ) {
+ description.mFormatID = kAudioFormatLinearPCM;
+ updateFormat = true;
+ }
+
+ if ( updateFormat ) {
+ result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &description );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+
+ // Now check the physical format.
+ property.mSelector = kAudioStreamPropertyPhysicalFormat;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ //std::cout << "Current physical stream format:" << std::endl;
+ //std::cout << " mBitsPerChan = " << description.mBitsPerChannel << std::endl;
+ //std::cout << " aligned high = " << (description.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (description.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
+ //std::cout << " bytesPerFrame = " << description.mBytesPerFrame << std::endl;
+ //std::cout << " sample rate = " << description.mSampleRate << std::endl;
+
+ if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 16 ) {
+ description.mFormatID = kAudioFormatLinearPCM;
+ //description.mSampleRate = (Float64) sampleRate;
+ AudioStreamBasicDescription testDescription = description;
+ UInt32 formatFlags;
+
+ // We'll try higher bit rates first and then work our way down.
+ std::vector< std::pair<UInt32, UInt32> > physicalFormats;
+ formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsFloat) & ~kLinearPCMFormatFlagIsSignedInteger;
+ physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
+ formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
+ physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
+ physicalFormats.push_back( std::pair<Float32, UInt32>( 24, formatFlags ) ); // 24-bit packed
+ formatFlags &= ~( kAudioFormatFlagIsPacked | kAudioFormatFlagIsAlignedHigh );
+ physicalFormats.push_back( std::pair<Float32, UInt32>( 24.2, formatFlags ) ); // 24-bit in 4 bytes, aligned low
+ formatFlags |= kAudioFormatFlagIsAlignedHigh;
+ physicalFormats.push_back( std::pair<Float32, UInt32>( 24.4, formatFlags ) ); // 24-bit in 4 bytes, aligned high
+ formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
+ physicalFormats.push_back( std::pair<Float32, UInt32>( 16, formatFlags ) );
+ physicalFormats.push_back( std::pair<Float32, UInt32>( 8, formatFlags ) );
+
+ bool setPhysicalFormat = false;
+ for( unsigned int i=0; i<physicalFormats.size(); i++ ) {
+ testDescription = description;
+ testDescription.mBitsPerChannel = (UInt32) physicalFormats[i].first;
+ testDescription.mFormatFlags = physicalFormats[i].second;
+ if ( (24 == (UInt32)physicalFormats[i].first) && ~( physicalFormats[i].second & kAudioFormatFlagIsPacked ) )
+ testDescription.mBytesPerFrame = 4 * testDescription.mChannelsPerFrame;
+ else
+ testDescription.mBytesPerFrame = testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;
+ testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;
+ result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &testDescription );
+ if ( result == noErr ) {
+ setPhysicalFormat = true;
+ //std::cout << "Updated physical stream format:" << std::endl;
+ //std::cout << " mBitsPerChan = " << testDescription.mBitsPerChannel << std::endl;
+ //std::cout << " aligned high = " << (testDescription.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (testDescription.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
+ //std::cout << " bytesPerFrame = " << testDescription.mBytesPerFrame << std::endl;
+ //std::cout << " sample rate = " << testDescription.mSampleRate << std::endl;
+ break;
+ }
+ }
+
+ if ( !setPhysicalFormat ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ } // done setting virtual/physical formats.
+
+ // Get the stream / device latency.
+ UInt32 latency;
+ dataSize = sizeof( UInt32 );
+ property.mSelector = kAudioDevicePropertyLatency;
+ if ( AudioObjectHasProperty( id, &property ) == true ) {
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &latency );
+ if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] = latency;
+ else {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting device latency for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ }
+ }
+
+ // Byte-swapping: According to AudioHardware.h, the stream data will
+ // always be presented in native-endian format, so we should never
+ // need to byte swap.
+ stream_.doByteSwap[mode] = false;
+
+ // From the CoreAudio documentation, PCM data must be supplied as
+ // 32-bit floats.
+ stream_.userFormat = format;
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+
+ if ( streamCount == 1 )
+ stream_.nDeviceChannels[mode] = description.mChannelsPerFrame;
+ else // multiple streams
+ stream_.nDeviceChannels[mode] = channels;
+ stream_.nUserChannels[mode] = channels;
+ stream_.channelOffset[mode] = channelOffset; // offset within a CoreAudio stream
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+ else stream_.userInterleaved = true;
+ stream_.deviceInterleaved[mode] = true;
+ if ( monoMode == true ) stream_.deviceInterleaved[mode] = false;
+
+ // Set flags for buffer conversion.
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( streamCount == 1 ) {
+ if ( stream_.nUserChannels[mode] > 1 &&
+ stream_.userInterleaved != stream_.deviceInterleaved[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ }
+ else if ( monoMode && stream_.userInterleaved )
+ stream_.doConvertBuffer[mode] = true;
+
+ // Allocate our CoreHandle structure for the stream.
+ CoreHandle *handle = 0;
+ if ( stream_.apiHandle == 0 ) {
+ try {
+ handle = new CoreHandle;
+ }
+ catch ( std::bad_alloc& ) {
+ errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory.";
+ goto error;
+ }
+
+ if ( pthread_cond_init( &handle->condition, NULL ) ) {
+ errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable.";
+ goto error;
+ }
+ stream_.apiHandle = (void *) handle;
+ }
+ else
+ handle = (CoreHandle *) stream_.apiHandle;
+ handle->iStream[mode] = firstStream;
+ handle->nStreams[mode] = streamCount;
+ handle->id[mode] = id;
+
+ // Allocate necessary internal buffers.
+ unsigned long bufferBytes;
+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ // stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ stream_.userBuffer[mode] = (char *) malloc( bufferBytes * sizeof(char) );
+ memset( stream_.userBuffer[mode], 0, bufferBytes * sizeof(char) );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
+ }
+
+ // If possible, we will make use of the CoreAudio stream buffers as
+ // "device buffers". However, we can't do this if using multiple
+ // streams.
+ if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) {
+
+ bool makeBuffer = true;
+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+ if ( mode == INPUT ) {
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ if ( bufferBytes <= bytesOut ) makeBuffer = false;
+ }
+ }
+
+ if ( makeBuffer ) {
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory.";
+ goto error;
+ }
+ }
+ }
+
+ stream_.sampleRate = sampleRate;
+ stream_.device[mode] = device;
+ stream_.state = STREAM_STOPPED;
+ stream_.callbackInfo.object = (void *) this;
+
+ // Setup the buffer conversion information structure.
+ if ( stream_.doConvertBuffer[mode] ) {
+ if ( streamCount > 1 ) setConvertInfo( mode, 0 );
+ else setConvertInfo( mode, channelOffset );
+ }
+
+ if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device )
+ // Only one callback procedure per device.
+ stream_.mode = DUPLEX;
+ else {
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+ result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] );
+#else
+ // deprecated in favor of AudioDeviceCreateIOProcID()
+ result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo );
+#endif
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ goto error;
+ }
+ if ( stream_.mode == OUTPUT && mode == INPUT )
+ stream_.mode = DUPLEX;
+ else
+ stream_.mode = mode;
+ }
+
+ // Setup the device property listener for over/underload.
+ property.mSelector = kAudioDeviceProcessorOverload;
+ property.mScope = kAudioObjectPropertyScopeGlobal;
+ result = AudioObjectAddPropertyListener( id, &property, xrunListener, (void *) handle );
+
+ return SUCCESS;
+
+ error:
+ if ( handle ) {
+ pthread_cond_destroy( &handle->condition );
+ delete handle;
+ stream_.apiHandle = 0;
+ }
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ stream_.state = STREAM_CLOSED;
+ return FAILURE;
+}
+
+void RtApiCore :: closeStream( void )
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiCore::closeStream(): no open stream to close!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ if (handle) {
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
+ kAudioObjectPropertyScopeGlobal,
+ kAudioObjectPropertyElementMaster };
+
+ property.mSelector = kAudioDeviceProcessorOverload;
+ property.mScope = kAudioObjectPropertyScopeGlobal;
+ if (AudioObjectRemovePropertyListener( handle->id[0], &property, xrunListener, (void *) handle ) != noErr) {
+ errorText_ = "RtApiCore::closeStream(): error removing property listener!";
+ error( RtAudioError::WARNING );
+ }
+ }
+ if ( stream_.state == STREAM_RUNNING )
+ AudioDeviceStop( handle->id[0], callbackHandler );
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+ AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] );
+#else
+ // deprecated in favor of AudioDeviceDestroyIOProcID()
+ AudioDeviceRemoveIOProc( handle->id[0], callbackHandler );
+#endif
+ }
+
+ if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
+ if (handle) {
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
+ kAudioObjectPropertyScopeGlobal,
+ kAudioObjectPropertyElementMaster };
+
+ property.mSelector = kAudioDeviceProcessorOverload;
+ property.mScope = kAudioObjectPropertyScopeGlobal;
+ if (AudioObjectRemovePropertyListener( handle->id[1], &property, xrunListener, (void *) handle ) != noErr) {
+ errorText_ = "RtApiCore::closeStream(): error removing property listener!";
+ error( RtAudioError::WARNING );
+ }
+ }
+ if ( stream_.state == STREAM_RUNNING )
+ AudioDeviceStop( handle->id[1], callbackHandler );
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+ AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] );
+#else
+ // deprecated in favor of AudioDeviceDestroyIOProcID()
+ AudioDeviceRemoveIOProc( handle->id[1], callbackHandler );
+#endif
+ }
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ // Destroy pthread condition variable.
+ pthread_cond_destroy( &handle->condition );
+ delete handle;
+ stream_.apiHandle = 0;
+
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
+}
+
+void RtApiCore :: startStream( void )
+{
+ verifyStream();
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiCore::startStream(): the stream is already running!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ OSStatus result = noErr;
+ CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+ result = AudioDeviceStart( handle->id[0], callbackHandler );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ if ( stream_.mode == INPUT ||
+ ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
+
+ result = AudioDeviceStart( handle->id[1], callbackHandler );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ handle->drainCounter = 0;
+ handle->internalDrain = false;
+ stream_.state = STREAM_RUNNING;
+
+ unlock:
+ if ( result == noErr ) return;
+ error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiCore :: stopStream( void )
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiCore::stopStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ OSStatus result = noErr;
+ CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+ if ( handle->drainCounter == 0 ) {
+ handle->drainCounter = 2;
+ pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
+ }
+
+ result = AudioDeviceStop( handle->id[0], callbackHandler );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
+
+ result = AudioDeviceStop( handle->id[1], callbackHandler );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ stream_.state = STREAM_STOPPED;
+
+ unlock:
+ if ( result == noErr ) return;
+ error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiCore :: abortStream( void )
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiCore::abortStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+ handle->drainCounter = 2;
+
+ 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 better to handle it this way because the
+// callbackEvent() function probably should return before the AudioDeviceStop()
+// function is called.
+static void *coreStopStream( void *ptr )
+{
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiCore *object = (RtApiCore *) info->object;
+
+ object->stopStream();
+ pthread_exit( NULL );
+}
+
+bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,
+ const AudioBufferList *inBufferList,
+ const AudioBufferList *outBufferList )
+{
+ if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtAudioError::WARNING );
+ return FAILURE;
+ }
+
+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+ CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+
+ // Check if we were draining the stream and signal is finished.
+ if ( handle->drainCounter > 3 ) {
+ ThreadHandle threadId;
+
+ stream_.state = STREAM_STOPPING;
+ if ( handle->internalDrain == true )
+ pthread_create( &threadId, NULL, coreStopStream, info );
+ else // external call to stopStream()
+ pthread_cond_signal( &handle->condition );
+ return SUCCESS;
+ }
+
+ AudioDeviceID outputDevice = handle->id[0];
+
+ // Invoke user callback to get fresh output data UNLESS we are
+ // draining stream or duplex mode AND the input/output devices are
+ // different AND this function is called for the input device.
+ if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) {
+ RtAudioCallback callback = (RtAudioCallback) info->callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
+ handle->xrun[0] = false;
+ }
+ if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
+ status |= RTAUDIO_INPUT_OVERFLOW;
+ handle->xrun[1] = false;
+ }
+
+ int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, info->userData );
+ if ( cbReturnValue == 2 ) {
+ stream_.state = STREAM_STOPPING;
+ handle->drainCounter = 2;
+ abortStream();
+ return SUCCESS;
+ }
+ else if ( cbReturnValue == 1 ) {
+ handle->drainCounter = 1;
+ handle->internalDrain = true;
+ }
+ }
+
+ if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) {
+
+ if ( handle->drainCounter > 1 ) { // write zeros to the output stream
+
+ if ( handle->nStreams[0] == 1 ) {
+ memset( outBufferList->mBuffers[handle->iStream[0]].mData,
+ 0,
+ outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
+ }
+ else { // fill multiple streams with zeros
+ for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
+ memset( outBufferList->mBuffers[handle->iStream[0]+i].mData,
+ 0,
+ outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize );
+ }
+ }
+ }
+ else if ( handle->nStreams[0] == 1 ) {
+ if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer
+ convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData,
+ stream_.userBuffer[0], stream_.convertInfo[0] );
+ }
+ else { // copy from user buffer
+ memcpy( outBufferList->mBuffers[handle->iStream[0]].mData,
+ stream_.userBuffer[0],
+ outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
+ }
+ }
+ else { // fill multiple streams
+ Float32 *inBuffer = (Float32 *) stream_.userBuffer[0];
+ if ( stream_.doConvertBuffer[0] ) {
+ convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+ inBuffer = (Float32 *) stream_.deviceBuffer;
+ }
+
+ if ( stream_.deviceInterleaved[0] == false ) { // mono mode
+ UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize;
+ for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
+ memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData,
+ (void *)&inBuffer[i*stream_.bufferSize], bufferBytes );
+ }
+ }
+ else { // fill multiple multi-channel streams with interleaved data
+ UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset;
+ Float32 *out, *in;
+
+ bool inInterleaved = ( stream_.userInterleaved ) ? true : false;
+ UInt32 inChannels = stream_.nUserChannels[0];
+ if ( stream_.doConvertBuffer[0] ) {
+ inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
+ inChannels = stream_.nDeviceChannels[0];
+ }
+
+ if ( inInterleaved ) inOffset = 1;
+ else inOffset = stream_.bufferSize;
+
+ channelsLeft = inChannels;
+ for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
+ in = inBuffer;
+ out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData;
+ streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels;
+
+ outJump = 0;
+ // Account for possible channel offset in first stream
+ if ( i == 0 && stream_.channelOffset[0] > 0 ) {
+ streamChannels -= stream_.channelOffset[0];
+ outJump = stream_.channelOffset[0];
+ out += outJump;
+ }
+
+ // Account for possible unfilled channels at end of the last stream
+ if ( streamChannels > channelsLeft ) {
+ outJump = streamChannels - channelsLeft;
+ streamChannels = channelsLeft;
+ }
+
+ // Determine input buffer offsets and skips
+ if ( inInterleaved ) {
+ inJump = inChannels;
+ in += inChannels - channelsLeft;
+ }
+ else {
+ inJump = 1;
+ in += (inChannels - channelsLeft) * inOffset;
+ }
+
+ for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
+ for ( unsigned int j=0; j<streamChannels; j++ ) {
+ *out++ = in[j*inOffset];
+ }
+ out += outJump;
+ in += inJump;
+ }
+ channelsLeft -= streamChannels;
+ }
+ }
+ }
+ }
+
+ // Don't bother draining input
+ if ( handle->drainCounter ) {
+ handle->drainCounter++;
+ goto unlock;
+ }
+
+ AudioDeviceID inputDevice;
+ inputDevice = handle->id[1];
+ if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) {
+
+ if ( handle->nStreams[1] == 1 ) {
+ if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer
+ convertBuffer( stream_.userBuffer[1],
+ (char *) inBufferList->mBuffers[handle->iStream[1]].mData,
+ stream_.convertInfo[1] );
+ }
+ else { // copy to user buffer
+ memcpy( stream_.userBuffer[1],
+ inBufferList->mBuffers[handle->iStream[1]].mData,
+ inBufferList->mBuffers[handle->iStream[1]].mDataByteSize );
+ }
+ }
+ else { // read from multiple streams
+ Float32 *outBuffer = (Float32 *) stream_.userBuffer[1];
+ if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer;
+
+ if ( stream_.deviceInterleaved[1] == false ) { // mono mode
+ UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize;
+ for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
+ memcpy( (void *)&outBuffer[i*stream_.bufferSize],
+ inBufferList->mBuffers[handle->iStream[1]+i].mData, bufferBytes );
+ }
+ }
+ else { // read from multiple multi-channel streams
+ UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset;
+ Float32 *out, *in;
+
+ bool outInterleaved = ( stream_.userInterleaved ) ? true : false;
+ UInt32 outChannels = stream_.nUserChannels[1];
+ if ( stream_.doConvertBuffer[1] ) {
+ outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
+ outChannels = stream_.nDeviceChannels[1];
+ }
+
+ if ( outInterleaved ) outOffset = 1;
+ else outOffset = stream_.bufferSize;
+
+ channelsLeft = outChannels;
+ for ( unsigned int i=0; i<handle->nStreams[1]; i++ ) {
+ out = outBuffer;
+ in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData;
+ streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels;
+
+ inJump = 0;
+ // Account for possible channel offset in first stream
+ if ( i == 0 && stream_.channelOffset[1] > 0 ) {
+ streamChannels -= stream_.channelOffset[1];
+ inJump = stream_.channelOffset[1];
+ in += inJump;
+ }
+
+ // Account for possible unread channels at end of the last stream
+ if ( streamChannels > channelsLeft ) {
+ inJump = streamChannels - channelsLeft;
+ streamChannels = channelsLeft;
+ }
+
+ // Determine output buffer offsets and skips
+ if ( outInterleaved ) {
+ outJump = outChannels;
+ out += outChannels - channelsLeft;
+ }
+ else {
+ outJump = 1;
+ out += (outChannels - channelsLeft) * outOffset;
+ }
+
+ for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
+ for ( unsigned int j=0; j<streamChannels; j++ ) {
+ out[j*outOffset] = *in++;
+ }
+ out += outJump;
+ in += inJump;
+ }
+ channelsLeft -= streamChannels;
+ }
+ }
+
+ if ( stream_.doConvertBuffer[1] ) { // convert from our internal "device" buffer
+ convertBuffer( stream_.userBuffer[1],
+ stream_.deviceBuffer,
+ stream_.convertInfo[1] );
+ }
+ }
+ }
+
+ unlock:
+ //MUTEX_UNLOCK( &stream_.mutex );
+
+ RtApi::tickStreamTime();
+ return SUCCESS;
+}
+
+const char* RtApiCore :: getErrorCode( OSStatus code )
+{
+ switch( code ) {
+
+ case kAudioHardwareNotRunningError:
+ return "kAudioHardwareNotRunningError";
+
+ case kAudioHardwareUnspecifiedError:
+ return "kAudioHardwareUnspecifiedError";
+
+ case kAudioHardwareUnknownPropertyError:
+ return "kAudioHardwareUnknownPropertyError";
+
+ case kAudioHardwareBadPropertySizeError:
+ return "kAudioHardwareBadPropertySizeError";
+
+ case kAudioHardwareIllegalOperationError:
+ return "kAudioHardwareIllegalOperationError";
+
+ case kAudioHardwareBadObjectError:
+ return "kAudioHardwareBadObjectError";
+
+ case kAudioHardwareBadDeviceError:
+ return "kAudioHardwareBadDeviceError";
+
+ case kAudioHardwareBadStreamError:
+ return "kAudioHardwareBadStreamError";
+
+ case kAudioHardwareUnsupportedOperationError:
+ return "kAudioHardwareUnsupportedOperationError";
+
+ case kAudioDeviceUnsupportedFormatError:
+ return "kAudioDeviceUnsupportedFormatError";
+
+ case kAudioDevicePermissionsError:
+ return "kAudioDevicePermissionsError";
+
+ default:
+ return "CoreAudio unknown error";
+ }
+}
+
+ //******************** End of __MACOSX_CORE__ *********************//
+#endif
+
+#if defined(__UNIX_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
+//
+// 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
+// RtApiJack 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>
+#include <unistd.h>
+#include <cstdio>
+
+// A structure to hold various information related to the Jack API
+// implementation.
+struct JackHandle {
+ jack_client_t *client;
+ jack_port_t **ports[2];
+ std::string deviceName[2];
+ bool xrun[2];
+ pthread_cond_t condition;
+ int drainCounter; // Tracks callback counts when draining
+ bool internalDrain; // Indicates if stop is initiated from callback or not.
+
+ JackHandle()
+ :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; }
+};
+
+static void jackSilentError( const char * ) {};
+
+RtApiJack :: RtApiJack()
+{
+ // Nothing to do here.
+#if !defined(__RTAUDIO_DEBUG__)
+ // Turn off Jack's internal error reporting.
+ jack_set_error_function( &jackSilentError );
+#endif
+}
+
+RtApiJack :: ~RtApiJack()
+{
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
+}
+
+unsigned int RtApiJack :: getDeviceCount( void )
+{
+ // 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( "RtApiJackCount", options, status );
+ if ( client == 0 ) return 0;
+
+ const char **ports;
+ std::string port, previousPort;
+ unsigned int 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 != std::string::npos ) {
+ port = port.substr( 0, iColon + 1 );
+ if ( port != previousPort ) {
+ nDevices++;
+ previousPort = port;
+ }
+ }
+ } while ( ports[++nChannels] );
+ free( ports );
+ }
+
+ jack_client_close( client );
+ return nDevices;
+}
+
+RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device )
+{
+ RtAudio::DeviceInfo info;
+ info.probed = false;
+
+ jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption
+ jack_status_t *status = NULL;
+ jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status );
+ if ( client == 0 ) {
+ errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!";
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ const char **ports;
+ std::string port, previousPort;
+ unsigned int nPorts = 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[ nPorts ];
+ iColon = port.find(":");
+ if ( iColon != std::string::npos ) {
+ port = port.substr( 0, iColon );
+ if ( port != previousPort ) {
+ if ( nDevices == device ) info.name = port;
+ nDevices++;
+ previousPort = port;
+ }
+ }
+ } while ( ports[++nPorts] );
+ free( ports );
+ }
+
+ if ( device >= nDevices ) {
+ jack_client_close( client );
+ errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";
+ error( RtAudioError::INVALID_USE );
+ return info;
+ }
+
+ // Get the current jack server sample rate.
+ info.sampleRates.clear();
+
+ info.preferredSampleRate = jack_get_sample_rate( client );
+ info.sampleRates.push_back( info.preferredSampleRate );
+
+ // Count the available ports containing the client name as device
+ // channels. Jack "input ports" equal RtAudio output channels.
+ unsigned int nChannels = 0;
+ ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput );
+ if ( ports ) {
+ while ( ports[ nChannels ] ) nChannels++;
+ free( ports );
+ info.outputChannels = nChannels;
+ }
+
+ // Jack "output ports" equal RtAudio input channels.
+ nChannels = 0;
+ ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput );
+ if ( ports ) {
+ while ( ports[ nChannels ] ) nChannels++;
+ free( ports );
+ info.inputChannels = nChannels;
+ }
+
+ if ( info.outputChannels == 0 && info.inputChannels == 0 ) {
+ jack_client_close(client);
+ errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";
+ error( RtAudioError::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 = RTAUDIO_FLOAT32;
+
+ // 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;
+
+ jack_client_close(client);
+ info.probed = true;
+ return info;
+}
+
+static int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )
+{
+ CallbackInfo *info = (CallbackInfo *) infoPointer;
+
+ RtApiJack *object = (RtApiJack *) info->object;
+ if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1;
+
+ return 0;
+}
+
+// This function will be called by a spawned thread when the Jack
+// server signals that it is shutting down. It is necessary to handle
+// it this way because the jackShutdown() function must return before
+// the jack_deactivate() function (in closeStream()) will return.
+static void *jackCloseStream( void *ptr )
+{
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiJack *object = (RtApiJack *) info->object;
+
+ object->closeStream();
+
+ pthread_exit( NULL );
+}
+static void jackShutdown( void *infoPointer )
+{
+ CallbackInfo *info = (CallbackInfo *) infoPointer;
+ RtApiJack *object = (RtApiJack *) info->object;
+
+ // Check current stream state. If stopped, then we'll assume this
+ // was called as a result of a call to RtApiJack::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 ( object->isStreamRunning() == false ) return;
+
+ ThreadHandle threadId;
+ pthread_create( &threadId, NULL, jackCloseStream, info );
+ std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;
+}
+
+static int jackXrun( void *infoPointer )
+{
+ JackHandle *handle = (JackHandle *) infoPointer;
+
+ if ( handle->ports[0] ) handle->xrun[0] = true;
+ if ( handle->ports[1] ) handle->xrun[1] = true;
+
+ return 0;
+}
+
+bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options )
+{
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+
+ // Look for jack server and try to become a client (only do once per stream).
+ jack_client_t *client = 0;
+ if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) {
+ jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
+ jack_status_t *status = NULL;
+ if ( options && !options->streamName.empty() )
+ client = jack_client_open( options->streamName.c_str(), jackoptions, status );
+ else
+ client = jack_client_open( "RtApiJack", jackoptions, status );
+ if ( client == 0 ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!";
+ error( RtAudioError::WARNING );
+ return FAILURE;
+ }
+ }
+ else {
+ // The handle must have been created on an earlier pass.
+ client = handle->client;
+ }
+
+ const char **ports;
+ std::string port, previousPort, deviceName;
+ unsigned int nPorts = 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[ nPorts ];
+ iColon = port.find(":");
+ if ( iColon != std::string::npos ) {
+ port = port.substr( 0, iColon );
+ if ( port != previousPort ) {
+ if ( nDevices == device ) deviceName = port;
+ nDevices++;
+ previousPort = port;
+ }
+ }
+ } while ( ports[++nPorts] );
+ free( ports );
+ }
+
+ if ( device >= nDevices ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!";
+ return FAILURE;
+ }
+
+ // Count the available ports containing the client name as device
+ // channels. Jack "input ports" equal RtAudio output channels.
+ unsigned int nChannels = 0;
+ unsigned long flag = JackPortIsInput;
+ if ( 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) ) {
+ errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Check the jack server sample rate.
+ unsigned int jackRate = jack_get_sample_rate( client );
+ if ( sampleRate != jackRate ) {
+ jack_client_close( client );
+ errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ stream_.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 == 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!
+ stream_.latency[mode] = latrange.min;
+ //stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );
+ }
+ free( ports );
+
+ // The jack server always uses 32-bit floating-point data.
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+ stream_.userFormat = format;
+
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+ else stream_.userInterleaved = true;
+
+ // Jack always uses non-interleaved buffers.
+ stream_.deviceInterleaved[mode] = false;
+
+ // Jack always provides host byte-ordered data.
+ stream_.doByteSwap[mode] = false;
+
+ // Get the buffer size. The buffer size and number of buffers
+ // (periods) is set when the jack server is started.
+ stream_.bufferSize = (int) jack_get_buffer_size( client );
+ *bufferSize = stream_.bufferSize;
+
+ stream_.nDeviceChannels[mode] = channels;
+ stream_.nUserChannels[mode] = channels;
+
+ // Set flags for buffer conversion.
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
+
+ // Allocate our JackHandle structure for the stream.
+ if ( handle == 0 ) {
+ try {
+ handle = new JackHandle;
+ }
+ catch ( std::bad_alloc& ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory.";
+ goto error;
+ }
+
+ if ( pthread_cond_init(&handle->condition, NULL) ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable.";
+ goto error;
+ }
+ stream_.apiHandle = (void *) handle;
+ handle->client = client;
+ }
+ handle->deviceName[mode] = deviceName;
+
+ // Allocate necessary internal buffers.
+ unsigned long bufferBytes;
+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
+ }
+
+ if ( stream_.doConvertBuffer[mode] ) {
+
+ bool makeBuffer = true;
+ if ( mode == OUTPUT )
+ bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ else { // mode == INPUT
+ bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] );
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
+ if ( bufferBytes < bytesOut ) makeBuffer = false;
+ }
+ }
+
+ if ( makeBuffer ) {
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory.";
+ goto error;
+ }
+ }
+ }
+
+ // Allocate memory for the Jack ports (channels) identifiers.
+ handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels );
+ if ( handle->ports[mode] == NULL ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory.";
+ goto error;
+ }
+
+ stream_.device[mode] = device;
+ stream_.channelOffset[mode] = firstChannel;
+ stream_.state = STREAM_STOPPED;
+ stream_.callbackInfo.object = (void *) this;
+
+ if ( stream_.mode == OUTPUT && mode == INPUT )
+ // We had already set up the stream for output.
+ stream_.mode = DUPLEX;
+ else {
+ stream_.mode = mode;
+ jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo );
+ jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle );
+ jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );
+ }
+
+ // Register our ports.
+ char label[64];
+ if ( mode == OUTPUT ) {
+ for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
+ snprintf( label, 64, "outport %d", i );
+ handle->ports[0][i] = jack_port_register( handle->client, (const char *)label,
+ JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 );
+ }
+ }
+ else {
+ for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
+ snprintf( label, 64, "inport %d", i );
+ handle->ports[1][i] = jack_port_register( handle->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 ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
+
+ return SUCCESS;
+
+ error:
+ if ( handle ) {
+ pthread_cond_destroy( &handle->condition );
+ jack_client_close( handle->client );
+
+ if ( handle->ports[0] ) free( handle->ports[0] );
+ if ( handle->ports[1] ) free( handle->ports[1] );
+
+ delete handle;
+ stream_.apiHandle = 0;
+ }
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ return FAILURE;
+}
+
+void RtApiJack :: closeStream( void )
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiJack::closeStream(): no open stream to close!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+ if ( handle ) {
+
+ if ( stream_.state == STREAM_RUNNING )
+ jack_deactivate( handle->client );
+
+ jack_client_close( handle->client );
+ }
+
+ if ( handle ) {
+ if ( handle->ports[0] ) free( handle->ports[0] );
+ if ( handle->ports[1] ) free( handle->ports[1] );
+ pthread_cond_destroy( &handle->condition );
+ delete handle;
+ stream_.apiHandle = 0;
+ }
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
+}
+
+void RtApiJack :: startStream( void )
+{
+ verifyStream();
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiJack::startStream(): the stream is already running!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+ int result = jack_activate( handle->client );
+ if ( result ) {
+ errorText_ = "RtApiJack::startStream(): unable to activate JACK client!";
+ goto unlock;
+ }
+
+ const char **ports;
+
+ // Get the list of available ports.
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ result = 1;
+ ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput);
+ if ( ports == NULL) {
+ errorText_ = "RtApiJack::startStream(): 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 ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
+ result = 1;
+ if ( ports[ stream_.channelOffset[0] + i ] )
+ result = jack_connect( handle->client, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] );
+ if ( result ) {
+ free( ports );
+ errorText_ = "RtApiJack::startStream(): error connecting output ports!";
+ goto unlock;
+ }
+ }
+ free(ports);
+ }
+
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+ result = 1;
+ ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput );
+ if ( ports == NULL) {
+ errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!";
+ goto unlock;
+ }
+
+ // Now make the port connections. See note above.
+ for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
+ result = 1;
+ if ( ports[ stream_.channelOffset[1] + i ] )
+ result = jack_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) );
+ if ( result ) {
+ free( ports );
+ errorText_ = "RtApiJack::startStream(): error connecting input ports!";
+ goto unlock;
+ }
+ }
+ free(ports);
+ }
+
+ handle->drainCounter = 0;
+ handle->internalDrain = false;
+ stream_.state = STREAM_RUNNING;
+
+ unlock:
+ if ( result == 0 ) return;
+ error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiJack :: stopStream( void )
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+ if ( handle->drainCounter == 0 ) {
+ handle->drainCounter = 2;
+ pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
+ }
+ }
+
+ jack_deactivate( handle->client );
+ stream_.state = STREAM_STOPPED;
+}
+
+void RtApiJack :: abortStream( void )
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+ handle->drainCounter = 2;
+
+ stopStream();
+}
+
+// This function will be called by a spawned thread when the user
+// callback function signals that the stream should be stopped or
+// aborted. It is necessary to handle it this way because the
+// callbackEvent() function must return before the jack_deactivate()
+// function will return.
+static void *jackStopStream( void *ptr )
+{
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiJack *object = (RtApiJack *) info->object;
+
+ object->stopStream();
+ pthread_exit( NULL );
+}
+
+bool RtApiJack :: callbackEvent( unsigned long nframes )
+{
+ if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtAudioError::WARNING );
+ return FAILURE;
+ }
+ if ( stream_.bufferSize != nframes ) {
+ errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";
+ error( RtAudioError::WARNING );
+ return FAILURE;
+ }
+
+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+
+ // Check if we were draining the stream and signal is finished.
+ if ( handle->drainCounter > 3 ) {
+ ThreadHandle threadId;
+
+ stream_.state = STREAM_STOPPING;
+ if ( handle->internalDrain == true )
+ pthread_create( &threadId, NULL, jackStopStream, info );
+ else
+ pthread_cond_signal( &handle->condition );
+ return SUCCESS;
+ }
+
+ // Invoke user callback first, to get fresh output data.
+ if ( handle->drainCounter == 0 ) {
+ RtAudioCallback callback = (RtAudioCallback) info->callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
+ handle->xrun[0] = false;
+ }
+ if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
+ status |= RTAUDIO_INPUT_OVERFLOW;
+ handle->xrun[1] = false;
+ }
+ int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, info->userData );
+ if ( cbReturnValue == 2 ) {
+ stream_.state = STREAM_STOPPING;
+ handle->drainCounter = 2;
+ ThreadHandle id;
+ pthread_create( &id, NULL, jackStopStream, info );
+ return SUCCESS;
+ }
+ else if ( cbReturnValue == 1 ) {
+ handle->drainCounter = 1;
+ handle->internalDrain = true;
+ }
+ }
+
+ jack_default_audio_sample_t *jackbuffer;
+ unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t );
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+ if ( handle->drainCounter > 1 ) { // write zeros to the output stream
+
+ for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
+ memset( jackbuffer, 0, bufferBytes );
+ }
+
+ }
+ else if ( stream_.doConvertBuffer[0] ) {
+
+ convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+
+ for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
+ memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes );
+ }
+ }
+ else { // no buffer conversion
+ for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
+ memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes );
+ }
+ }
+ }
+
+ // Don't bother draining input
+ if ( handle->drainCounter ) {
+ handle->drainCounter++;
+ goto unlock;
+ }
+
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+
+ if ( stream_.doConvertBuffer[1] ) {
+ for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) {
+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
+ memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes );
+ }
+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+ }
+ else { // no buffer conversion
+ for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
+ memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes );
+ }
+ }
+ }
+
+ unlock:
+ RtApi::tickStreamTime();
+ return SUCCESS;
+}
+ //******************** End of __UNIX_JACK__ *********************//
+#endif
+
+#if defined(__WINDOWS_ASIO__) // ASIO API on Windows
+
+// 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.
+
+#include "asiosys.h"
+#include "asio.h"
+#include "iasiothiscallresolver.h"
+#include "asiodrivers.h"
+#include <cmath>
+
+static AsioDrivers drivers;
+static ASIOCallbacks asioCallbacks;
+static ASIODriverInfo driverInfo;
+static CallbackInfo *asioCallbackInfo;
+static bool asioXRun;
+
+struct AsioHandle {
+ int drainCounter; // Tracks callback counts when draining
+ bool internalDrain; // Indicates if stop is initiated from callback or not.
+ ASIOBufferInfo *bufferInfos;
+ HANDLE condition;
+
+ AsioHandle()
+ :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 );
+
+RtApiAsio :: RtApiAsio()
+{
+ // 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).
+ coInitialized_ = false;
+ HRESULT hr = CoInitialize( NULL );
+ if ( FAILED(hr) ) {
+ errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";
+ error( RtAudioError::WARNING );
+ }
+ coInitialized_ = true;
+
+ drivers.removeCurrentDriver();
+ driverInfo.asioVersion = 2;
+
+ // See note in DirectSound implementation about GetDesktopWindow().
+ driverInfo.sysRef = GetForegroundWindow();
+}
+
+RtApiAsio :: ~RtApiAsio()
+{
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
+ if ( coInitialized_ ) CoUninitialize();
+}
+
+unsigned int RtApiAsio :: getDeviceCount( void )
+{
+ return (unsigned int) drivers.asioGetNumDev();
+}
+
+RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device )
+{
+ RtAudio::DeviceInfo info;
+ info.probed = false;
+
+ // Get device ID
+ unsigned int nDevices = getDeviceCount();
+ if ( nDevices == 0 ) {
+ errorText_ = "RtApiAsio::getDeviceInfo: no devices found!";
+ error( RtAudioError::INVALID_USE );
+ return info;
+ }
+
+ if ( device >= nDevices ) {
+ errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";
+ error( RtAudioError::INVALID_USE );
+ return info;
+ }
+
+ // If a stream is already open, we cannot probe other devices. Thus, use the saved results.
+ if ( stream_.state != STREAM_CLOSED ) {
+ if ( device >= devices_.size() ) {
+ errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened.";
+ error( RtAudioError::WARNING );
+ return info;
+ }
+ return devices_[ device ];
+ }
+
+ char driverName[32];
+ ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ info.name = driverName;
+
+ if ( !drivers.loadDriver( driverName ) ) {
+ errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ result = ASIOInit( &driverInfo );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // Determine the device channel information.
+ long inputChannels, outputChannels;
+ result = ASIOGetChannels( &inputChannels, &outputChannels );
+ if ( result != ASE_OK ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ 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 ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
+ result = ASIOCanSampleRate( (ASIOSampleRate) SAMPLE_RATES[i] );
+ if ( result == ASE_OK ) {
+ info.sampleRates.push_back( SAMPLE_RATES[i] );
+
+ if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
+ info.preferredSampleRate = 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();
+ errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ info.nativeFormats = 0;
+ if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB )
+ info.nativeFormats |= RTAUDIO_SINT16;
+ else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB )
+ info.nativeFormats |= RTAUDIO_SINT32;
+ else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB )
+ info.nativeFormats |= RTAUDIO_FLOAT32;
+ else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB )
+ info.nativeFormats |= RTAUDIO_FLOAT64;
+ else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB )
+ info.nativeFormats |= RTAUDIO_SINT24;
+
+ 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 RtApiAsio :: saveDeviceInfo( void )
+{
+ devices_.clear();
+
+ unsigned int nDevices = getDeviceCount();
+ devices_.resize( nDevices );
+ for ( unsigned int i=0; i<nDevices; i++ )
+ devices_[i] = getDeviceInfo( i );
+}
+
+bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options )
+{////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
+
+ bool isDuplexInput = mode == INPUT && stream_.mode == OUTPUT;
+
+ // For ASIO, a duplex stream MUST use the same driver.
+ if ( isDuplexInput && stream_.device[0] != device ) {
+ errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!";
+ return FAILURE;
+ }
+
+ char driverName[32];
+ ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: unable to get driver name (" << getAsioErrorString( result ) << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Only load the driver once for duplex stream.
+ if ( !isDuplexInput ) {
+ // 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 ) ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ result = ASIOInit( &driverInfo );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+
+ // keep them before any "goto error", they are used for error cleanup + goto device boundary checks
+ bool buffersAllocated = false;
+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ unsigned int nChannels;
+
+
+ // Check the device channel count.
+ long inputChannels, outputChannels;
+ result = ASIOGetChannels( &inputChannels, &outputChannels );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ goto error;
+ }
+
+ if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) ||
+ ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ").";
+ errorText_ = errorStream_.str();
+ goto error;
+ }
+ stream_.nDeviceChannels[mode] = channels;
+ stream_.nUserChannels[mode] = channels;
+ stream_.channelOffset[mode] = firstChannel;
+
+ // Verify the sample rate is supported.
+ result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ").";
+ errorText_ = errorStream_.str();
+ goto error;
+ }
+
+ // Get the current sample rate
+ ASIOSampleRate currentRate;
+ result = ASIOGetSampleRate( ¤tRate );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";
+ errorText_ = errorStream_.str();
+ goto error;
+ }
+
+ // Set the sample rate only if necessary
+ if ( currentRate != sampleRate ) {
+ result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";
+ errorText_ = errorStream_.str();
+ goto error;
+ }
+ }
+
+ // Determine the driver data type.
+ ASIOChannelInfo channelInfo;
+ channelInfo.channel = 0;
+ if ( mode == OUTPUT ) channelInfo.isInput = false;
+ else channelInfo.isInput = true;
+ result = ASIOGetChannelInfo( &channelInfo );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format.";
+ errorText_ = errorStream_.str();
+ goto error;
+ }
+
+ // Assuming WINDOWS host is always little-endian.
+ stream_.doByteSwap[mode] = false;
+ stream_.userFormat = format;
+ stream_.deviceFormat[mode] = 0;
+ if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true;
+ }
+ else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true;
+ }
+ else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) {
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+ if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true;
+ }
+ else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) {
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
+ if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true;
+ }
+ else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+ if ( channelInfo.type == ASIOSTInt24MSB ) stream_.doByteSwap[mode] = true;
+ }
+
+ if ( stream_.deviceFormat[mode] == 0 ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";
+ errorText_ = errorStream_.str();
+ goto error;
+ }
+
+ // 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 ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size.";
+ errorText_ = errorStream_.str();
+ goto error;
+ }
+
+ if ( isDuplexInput ) {
+ // When this is the duplex input (output was opened before), then we have to use the same
+ // buffersize as the output, because it might use the preferred buffer size, which most
+ // likely wasn't passed as input to this. The buffer sizes have to be identically anyway,
+ // So instead of throwing an error, make them equal. The caller uses the reference
+ // to the "bufferSize" param as usual to set up processing buffers.
+
+ *bufferSize = stream_.bufferSize;
+
+ } else {
+ if ( *bufferSize == 0 ) *bufferSize = preferSize;
+ else if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
+ else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
+ else if ( granularity == -1 ) {
+ // Make sure bufferSize is a power of two.
+ int log2_of_min_size = 0;
+ int log2_of_max_size = 0;
+
+ for ( unsigned int 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 = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) );
+ int min_delta_num = log2_of_min_size;
+
+ for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) {
+ long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) );
+ if (current_delta < min_delta) {
+ min_delta = current_delta;
+ min_delta_num = i;
+ }
+ }
+
+ *bufferSize = ( (unsigned int)1 << min_delta_num );
+ if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
+ else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
+ }
+ else if ( granularity != 0 ) {
+ // Set to an even multiple of granularity, rounding up.
+ *bufferSize = (*bufferSize + granularity-1) / granularity * granularity;
+ }
+ }
+
+ /*
+ // we don't use it anymore, see above!
+ // Just left it here for the case...
+ if ( isDuplexInput && stream_.bufferSize != *bufferSize ) {
+ errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!";
+ goto error;
+ }
+ */
+
+ stream_.bufferSize = *bufferSize;
+ stream_.nBuffers = 2;
+
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+ else stream_.userInterleaved = true;
+
+ // ASIO always uses non-interleaved buffers.
+ stream_.deviceInterleaved[mode] = false;
+
+ // Allocate, if necessary, our AsioHandle structure for the stream.
+ if ( handle == 0 ) {
+ try {
+ handle = new AsioHandle;
+ }
+ catch ( std::bad_alloc& ) {
+ errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory.";
+ goto error;
+ }
+ handle->bufferInfos = 0;
+
+ // Create a manual-reset event.
+ handle->condition = CreateEvent( NULL, // no security
+ TRUE, // manual-reset
+ FALSE, // non-signaled initially
+ NULL ); // unnamed
+ stream_.apiHandle = (void *) handle;
+ }
+
+ // 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.
+ if ( mode == INPUT && stream_.mode == OUTPUT ) {
+ ASIODisposeBuffers();
+ if ( handle->bufferInfos ) free( handle->bufferInfos );
+ }
+
+ // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
+ unsigned int i;
+ nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
+ handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) );
+ if ( handle->bufferInfos == NULL ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ goto error;
+ }
+
+ ASIOBufferInfo *infos;
+ infos = handle->bufferInfos;
+ for ( i=0; i<stream_.nDeviceChannels[0]; i++, infos++ ) {
+ infos->isInput = ASIOFalse;
+ infos->channelNum = i + stream_.channelOffset[0];
+ infos->buffers[0] = infos->buffers[1] = 0;
+ }
+ for ( i=0; i<stream_.nDeviceChannels[1]; i++, infos++ ) {
+ infos->isInput = ASIOTrue;
+ infos->channelNum = i + stream_.channelOffset[1];
+ infos->buffers[0] = infos->buffers[1] = 0;
+ }
+
+ // prepare for callbacks
+ stream_.sampleRate = sampleRate;
+ stream_.device[mode] = device;
+ stream_.mode = isDuplexInput ? DUPLEX : mode;
+
+ // store this class instance before registering callbacks, that are going to use it
+ asioCallbackInfo = &stream_.callbackInfo;
+ stream_.callbackInfo.object = (void *) this;
+
+ // 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( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
+ if ( result != ASE_OK ) {
+ // Standard method failed. This can happen with strict/misbehaving drivers that return valid buffer size ranges
+ // but only accept the preferred buffer size as parameter for ASIOCreateBuffers. eg. Creatives ASIO driver
+ // in that case, let's be naïve and try that instead
+ *bufferSize = preferSize;
+ stream_.bufferSize = *bufferSize;
+ result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
+ }
+
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers.";
+ errorText_ = errorStream_.str();
+ goto error;
+ }
+ buffersAllocated = true;
+ stream_.state = STREAM_STOPPED;
+
+ // Set flags for buffer conversion.
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
+
+ // Allocate necessary internal buffers
+ unsigned long bufferBytes;
+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
+ }
+
+ if ( stream_.doConvertBuffer[mode] ) {
+
+ bool makeBuffer = true;
+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+ if ( isDuplexInput && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ if ( bufferBytes <= bytesOut ) makeBuffer = false;
+ }
+
+ if ( makeBuffer ) {
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory.";
+ goto error;
+ }
+ }
+ }
+
+ // Determine device latencies
+ long inputLatency, outputLatency;
+ result = ASIOGetLatencies( &inputLatency, &outputLatency );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING); // warn but don't fail
+ }
+ else {
+ stream_.latency[0] = outputLatency;
+ stream_.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 ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
+
+ return SUCCESS;
+
+ error:
+ if ( !isDuplexInput ) {
+ // the cleanup for error in the duplex input, is done by RtApi::openStream
+ // So we clean up for single channel only
+
+ if ( buffersAllocated )
+ ASIODisposeBuffers();
+
+ drivers.removeCurrentDriver();
+
+ if ( handle ) {
+ CloseHandle( handle->condition );
+ if ( handle->bufferInfos )
+ free( handle->bufferInfos );
+
+ delete handle;
+ stream_.apiHandle = 0;
+ }
+
+
+ if ( stream_.userBuffer[mode] ) {
+ free( stream_.userBuffer[mode] );
+ stream_.userBuffer[mode] = 0;
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+ }
+
+ return FAILURE;
+}////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
+
+void RtApiAsio :: closeStream()
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiAsio::closeStream(): no open stream to close!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ if ( stream_.state == STREAM_RUNNING ) {
+ stream_.state = STREAM_STOPPED;
+ ASIOStop();
+ }
+ ASIODisposeBuffers();
+ drivers.removeCurrentDriver();
+
+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ if ( handle ) {
+ CloseHandle( handle->condition );
+ if ( handle->bufferInfos )
+ free( handle->bufferInfos );
+ delete handle;
+ stream_.apiHandle = 0;
+ }
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
+}
+
+bool stopThreadCalled = false;
+
+void RtApiAsio :: startStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiAsio::startStream(): the stream is already running!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ ASIOError result = ASIOStart();
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device.";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+
+ handle->drainCounter = 0;
+ handle->internalDrain = false;
+ ResetEvent( handle->condition );
+ stream_.state = STREAM_RUNNING;
+ asioXRun = false;
+
+ unlock:
+ stopThreadCalled = false;
+
+ if ( result == ASE_OK ) return;
+ error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiAsio :: stopStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ if ( handle->drainCounter == 0 ) {
+ handle->drainCounter = 2;
+ WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
+ }
+ }
+
+ stream_.state = STREAM_STOPPED;
+
+ ASIOError result = ASIOStop();
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device.";
+ errorText_ = errorStream_.str();
+ }
+
+ if ( result == ASE_OK ) return;
+ error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiAsio :: abortStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";
+ error( RtAudioError::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.
+ // AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ // handle->drainCounter = 2;
+ 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 RtApiAsio :: callbackEvent( long bufferIndex )
+{
+ if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtAudioError::WARNING );
+ return FAILURE;
+ }
+
+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+
+ // Check if we were draining the stream and signal if finished.
+ if ( handle->drainCounter > 3 ) {
+
+ stream_.state = STREAM_STOPPING;
+ if ( handle->internalDrain == false )
+ SetEvent( handle->condition );
+ else { // spawn a thread to stop the stream
+ unsigned threadId;
+ stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
+ &stream_.callbackInfo, 0, &threadId );
+ }
+ return SUCCESS;
+ }
+
+ // Invoke user callback to get fresh output data UNLESS we are
+ // draining stream.
+ if ( handle->drainCounter == 0 ) {
+ RtAudioCallback callback = (RtAudioCallback) info->callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ if ( stream_.mode != INPUT && asioXRun == true ) {
+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
+ asioXRun = false;
+ }
+ if ( stream_.mode != OUTPUT && asioXRun == true ) {
+ status |= RTAUDIO_INPUT_OVERFLOW;
+ asioXRun = false;
+ }
+ int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, info->userData );
+ if ( cbReturnValue == 2 ) {
+ stream_.state = STREAM_STOPPING;
+ handle->drainCounter = 2;
+ unsigned threadId;
+ stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
+ &stream_.callbackInfo, 0, &threadId );
+ return SUCCESS;
+ }
+ else if ( cbReturnValue == 1 ) {
+ handle->drainCounter = 1;
+ handle->internalDrain = true;
+ }
+ }
+
+ unsigned int nChannels, bufferBytes, i, j;
+ nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+ bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] );
+
+ if ( handle->drainCounter > 1 ) { // write zeros to the output stream
+
+ for ( i=0, j=0; i<nChannels; i++ ) {
+ if ( handle->bufferInfos[i].isInput != ASIOTrue )
+ memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes );
+ }
+
+ }
+ else if ( stream_.doConvertBuffer[0] ) {
+
+ convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+ if ( stream_.doByteSwap[0] )
+ byteSwapBuffer( stream_.deviceBuffer,
+ stream_.bufferSize * stream_.nDeviceChannels[0],
+ stream_.deviceFormat[0] );
+
+ for ( i=0, j=0; i<nChannels; i++ ) {
+ if ( handle->bufferInfos[i].isInput != ASIOTrue )
+ memcpy( handle->bufferInfos[i].buffers[bufferIndex],
+ &stream_.deviceBuffer[j++*bufferBytes], bufferBytes );
+ }
+
+ }
+ else {
+
+ if ( stream_.doByteSwap[0] )
+ byteSwapBuffer( stream_.userBuffer[0],
+ stream_.bufferSize * stream_.nUserChannels[0],
+ stream_.userFormat );
+
+ for ( i=0, j=0; i<nChannels; i++ ) {
+ if ( handle->bufferInfos[i].isInput != ASIOTrue )
+ memcpy( handle->bufferInfos[i].buffers[bufferIndex],
+ &stream_.userBuffer[0][bufferBytes*j++], bufferBytes );
+ }
+
+ }
+ }
+
+ // Don't bother draining input
+ if ( handle->drainCounter ) {
+ handle->drainCounter++;
+ goto unlock;
+ }
+
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+
+ bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]);
+
+ if (stream_.doConvertBuffer[1]) {
+
+ // Always interleave ASIO input data.
+ for ( i=0, j=0; i<nChannels; i++ ) {
+ if ( handle->bufferInfos[i].isInput == ASIOTrue )
+ memcpy( &stream_.deviceBuffer[j++*bufferBytes],
+ handle->bufferInfos[i].buffers[bufferIndex],
+ bufferBytes );
+ }
+
+ if ( stream_.doByteSwap[1] )
+ byteSwapBuffer( stream_.deviceBuffer,
+ stream_.bufferSize * stream_.nDeviceChannels[1],
+ stream_.deviceFormat[1] );
+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+
+ }
+ else {
+ for ( i=0, j=0; i<nChannels; i++ ) {
+ if ( handle->bufferInfos[i].isInput == ASIOTrue ) {
+ memcpy( &stream_.userBuffer[1][bufferBytes*j++],
+ handle->bufferInfos[i].buffers[bufferIndex],
+ bufferBytes );
+ }
+ }
+
+ if ( stream_.doByteSwap[1] )
+ byteSwapBuffer( stream_.userBuffer[1],
+ stream_.bufferSize * stream_.nUserChannels[1],
+ stream_.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();
+
+ RtApi::tickStreamTime();
+ return SUCCESS;
+}
+
+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;
+ try {
+ object->stopStream();
+ }
+ catch ( RtAudioError &exception ) {
+ std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl;
+ return;
+ }
+
+ std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl;
+}
+
+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.
+ std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl;
+ 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.
+ // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl;
+ 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.
+ std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl;
+ 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 ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i )
+ if ( m[i].value == result ) return m[i].message;
+
+ return "Unknown error.";
+}
+
+//******************** End of __WINDOWS_ASIO__ *********************//
+#endif
+
+
+#if defined(__WINDOWS_WASAPI__) // Windows WASAPI API
+
+// Authored by Marcus Tomlinson <themarcustomlinson at gmail.com>, April 2014
+// - Introduces support for the Windows WASAPI API
+// - Aims to deliver bit streams to and from hardware at the lowest possible latency, via the absolute minimum buffer sizes required
+// - Provides flexible stream configuration to an otherwise strict and inflexible WASAPI interface
+// - Includes automatic internal conversion of sample rate and buffer size between hardware and the user
+
+#ifndef INITGUID
+ #define INITGUID
+#endif
+#include <audioclient.h>
+#include <avrt.h>
+#include <mmdeviceapi.h>
+#include <functiondiscoverykeys_devpkey.h>
+
+//=============================================================================
+
+#define SAFE_RELEASE( objectPtr )\
+if ( objectPtr )\
+{\
+ objectPtr->Release();\
+ objectPtr = NULL;\
+}
+
+typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex );
+
+//-----------------------------------------------------------------------------
+
+// WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size.
+// Therefore we must perform all necessary conversions to user buffers in order to satisfy these
+// requirements. WasapiBuffer ring buffers are used between HwIn->UserIn and UserOut->HwOut to
+// provide intermediate storage for read / write synchronization.
+class WasapiBuffer
+{
+public:
+ WasapiBuffer()
+ : buffer_( NULL ),
+ bufferSize_( 0 ),
+ inIndex_( 0 ),
+ outIndex_( 0 ) {}
+
+ ~WasapiBuffer() {
+ free( buffer_ );
+ }
+
+ // sets the length of the internal ring buffer
+ void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) {
+ free( buffer_ );
+
+ buffer_ = ( char* ) calloc( bufferSize, formatBytes );
+
+ bufferSize_ = bufferSize;
+ inIndex_ = 0;
+ outIndex_ = 0;
+ }
+
+ // attempt to push a buffer into the ring buffer at the current "in" index
+ bool pushBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
+ {
+ if ( !buffer || // incoming buffer is NULL
+ bufferSize == 0 || // incoming buffer has no data
+ bufferSize > bufferSize_ ) // incoming buffer too large
+ {
+ return false;
+ }
+
+ unsigned int relOutIndex = outIndex_;
+ unsigned int inIndexEnd = inIndex_ + bufferSize;
+ if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) {
+ relOutIndex += bufferSize_;
+ }
+
+ // "in" index can end on the "out" index but cannot begin at it
+ if ( inIndex_ <= relOutIndex && inIndexEnd > relOutIndex ) {
+ return false; // not enough space between "in" index and "out" index
+ }
+
+ // copy buffer from external to internal
+ int fromZeroSize = inIndex_ + bufferSize - bufferSize_;
+ fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
+ int fromInSize = bufferSize - fromZeroSize;
+
+ switch( format )
+ {
+ case RTAUDIO_SINT8:
+ memcpy( &( ( char* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( char ) );
+ memcpy( buffer_, &( ( char* ) buffer )[fromInSize], fromZeroSize * sizeof( char ) );
+ break;
+ case RTAUDIO_SINT16:
+ memcpy( &( ( short* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( short ) );
+ memcpy( buffer_, &( ( short* ) buffer )[fromInSize], fromZeroSize * sizeof( short ) );
+ break;
+ case RTAUDIO_SINT24:
+ memcpy( &( ( S24* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( S24 ) );
+ memcpy( buffer_, &( ( S24* ) buffer )[fromInSize], fromZeroSize * sizeof( S24 ) );
+ break;
+ case RTAUDIO_SINT32:
+ memcpy( &( ( int* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( int ) );
+ memcpy( buffer_, &( ( int* ) buffer )[fromInSize], fromZeroSize * sizeof( int ) );
+ break;
+ case RTAUDIO_FLOAT32:
+ memcpy( &( ( float* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( float ) );
+ memcpy( buffer_, &( ( float* ) buffer )[fromInSize], fromZeroSize * sizeof( float ) );
+ break;
+ case RTAUDIO_FLOAT64:
+ memcpy( &( ( double* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( double ) );
+ memcpy( buffer_, &( ( double* ) buffer )[fromInSize], fromZeroSize * sizeof( double ) );
+ break;
+ }
+
+ // update "in" index
+ inIndex_ += bufferSize;
+ inIndex_ %= bufferSize_;
+
+ return true;
+ }
+
+ // attempt to pull a buffer from the ring buffer from the current "out" index
+ bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
+ {
+ if ( !buffer || // incoming buffer is NULL
+ bufferSize == 0 || // incoming buffer has no data
+ bufferSize > bufferSize_ ) // incoming buffer too large
+ {
+ return false;
+ }
+
+ unsigned int relInIndex = inIndex_;
+ unsigned int outIndexEnd = outIndex_ + bufferSize;
+ if ( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ ) {
+ relInIndex += bufferSize_;
+ }
+
+ // "out" index can begin at and end on the "in" index
+ if ( outIndex_ < relInIndex && outIndexEnd > relInIndex ) {
+ return false; // not enough space between "out" index and "in" index
+ }
+
+ // copy buffer from internal to external
+ int fromZeroSize = outIndex_ + bufferSize - bufferSize_;
+ fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
+ int fromOutSize = bufferSize - fromZeroSize;
+
+ switch( format )
+ {
+ case RTAUDIO_SINT8:
+ memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) );
+ memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) );
+ break;
+ case RTAUDIO_SINT16:
+ memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) );
+ memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) );
+ break;
+ case RTAUDIO_SINT24:
+ memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) );
+ memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) );
+ break;
+ case RTAUDIO_SINT32:
+ memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) );
+ memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) );
+ break;
+ case RTAUDIO_FLOAT32:
+ memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) );
+ memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) );
+ break;
+ case RTAUDIO_FLOAT64:
+ memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) );
+ memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) );
+ break;
+ }
+
+ // update "out" index
+ outIndex_ += bufferSize;
+ outIndex_ %= bufferSize_;
+
+ return true;
+ }
+
+private:
+ char* buffer_;
+ unsigned int bufferSize_;
+ unsigned int inIndex_;
+ unsigned int outIndex_;
+};
+
+//-----------------------------------------------------------------------------
+
+// In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate
+// between HW and the user. The convertBufferWasapi function is used to perform this conversion
+// between HwIn->UserIn and UserOut->HwOut during the stream callback loop.
+// This sample rate converter favors speed over quality, and works best with conversions between
+// one rate and its multiple.
+void convertBufferWasapi( char* outBuffer,
+ const char* inBuffer,
+ const unsigned int& channelCount,
+ const unsigned int& inSampleRate,
+ const unsigned int& outSampleRate,
+ const unsigned int& inSampleCount,
+ unsigned int& outSampleCount,
+ const RtAudioFormat& format )
+{
+ // calculate the new outSampleCount and relative sampleStep
+ float sampleRatio = ( float ) outSampleRate / inSampleRate;
+ float sampleStep = 1.0f / sampleRatio;
+ float inSampleFraction = 0.0f;
+
+ outSampleCount = ( unsigned int ) roundf( inSampleCount * sampleRatio );
+
+ // frame-by-frame, copy each relative input sample into it's corresponding output sample
+ for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )
+ {
+ unsigned int inSample = ( unsigned int ) inSampleFraction;
+
+ switch ( format )
+ {
+ case RTAUDIO_SINT8:
+ memcpy( &( ( char* ) outBuffer )[ outSample * channelCount ], &( ( char* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( char ) );
+ break;
+ case RTAUDIO_SINT16:
+ memcpy( &( ( short* ) outBuffer )[ outSample * channelCount ], &( ( short* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( short ) );
+ break;
+ case RTAUDIO_SINT24:
+ memcpy( &( ( S24* ) outBuffer )[ outSample * channelCount ], &( ( S24* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( S24 ) );
+ break;
+ case RTAUDIO_SINT32:
+ memcpy( &( ( int* ) outBuffer )[ outSample * channelCount ], &( ( int* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( int ) );
+ break;
+ case RTAUDIO_FLOAT32:
+ memcpy( &( ( float* ) outBuffer )[ outSample * channelCount ], &( ( float* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( float ) );
+ break;
+ case RTAUDIO_FLOAT64:
+ memcpy( &( ( double* ) outBuffer )[ outSample * channelCount ], &( ( double* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( double ) );
+ break;
+ }
+
+ // jump to next in sample
+ inSampleFraction += sampleStep;
+ }
+}
+
+//-----------------------------------------------------------------------------
+
+// A structure to hold various information related to the WASAPI implementation.
+struct WasapiHandle
+{
+ IAudioClient* captureAudioClient;
+ IAudioClient* renderAudioClient;
+ IAudioCaptureClient* captureClient;
+ IAudioRenderClient* renderClient;
+ HANDLE captureEvent;
+ HANDLE renderEvent;
+
+ WasapiHandle()
+ : captureAudioClient( NULL ),
+ renderAudioClient( NULL ),
+ captureClient( NULL ),
+ renderClient( NULL ),
+ captureEvent( NULL ),
+ renderEvent( NULL ) {}
+};
+
+//=============================================================================
+
+RtApiWasapi::RtApiWasapi()
+ : coInitialized_( false ), deviceEnumerator_( NULL )
+{
+ // WASAPI can run either apartment or multi-threaded
+ HRESULT hr = CoInitialize( NULL );
+ if ( !FAILED( hr ) )
+ coInitialized_ = true;
+
+ // Instantiate device enumerator
+ hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL,
+ CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ),
+ ( void** ) &deviceEnumerator_ );
+
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::RtApiWasapi: Unable to instantiate device enumerator";
+ error( RtAudioError::DRIVER_ERROR );
+ }
+}
+
+//-----------------------------------------------------------------------------
+
+RtApiWasapi::~RtApiWasapi()
+{
+ if ( stream_.state != STREAM_CLOSED )
+ closeStream();
+
+ SAFE_RELEASE( deviceEnumerator_ );
+
+ // If this object previously called CoInitialize()
+ if ( coInitialized_ )
+ CoUninitialize();
+}
+
+//=============================================================================
+
+unsigned int RtApiWasapi::getDeviceCount( void )
+{
+ unsigned int captureDeviceCount = 0;
+ unsigned int renderDeviceCount = 0;
+
+ IMMDeviceCollection* captureDevices = NULL;
+ IMMDeviceCollection* renderDevices = NULL;
+
+ // Count capture devices
+ errorText_.clear();
+ HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection.";
+ goto Exit;
+ }
+
+ hr = captureDevices->GetCount( &captureDeviceCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count.";
+ goto Exit;
+ }
+
+ // Count render devices
+ hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device collection.";
+ goto Exit;
+ }
+
+ hr = renderDevices->GetCount( &renderDeviceCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count.";
+ goto Exit;
+ }
+
+Exit:
+ // release all references
+ SAFE_RELEASE( captureDevices );
+ SAFE_RELEASE( renderDevices );
+
+ if ( errorText_.empty() )
+ return captureDeviceCount + renderDeviceCount;
+
+ error( RtAudioError::DRIVER_ERROR );
+ return 0;
+}
+
+//-----------------------------------------------------------------------------
+
+RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device )
+{
+ RtAudio::DeviceInfo info;
+ unsigned int captureDeviceCount = 0;
+ unsigned int renderDeviceCount = 0;
+ std::string defaultDeviceName;
+ bool isCaptureDevice = false;
+
+ PROPVARIANT deviceNameProp;
+ PROPVARIANT defaultDeviceNameProp;
+
+ IMMDeviceCollection* captureDevices = NULL;
+ IMMDeviceCollection* renderDevices = NULL;
+ IMMDevice* devicePtr = NULL;
+ IMMDevice* defaultDevicePtr = NULL;
+ IAudioClient* audioClient = NULL;
+ IPropertyStore* devicePropStore = NULL;
+ IPropertyStore* defaultDevicePropStore = NULL;
+
+ WAVEFORMATEX* deviceFormat = NULL;
+ WAVEFORMATEX* closestMatchFormat = NULL;
+
+ // probed
+ info.probed = false;
+
+ // Count capture devices
+ errorText_.clear();
+ RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
+ HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device collection.";
+ goto Exit;
+ }
+
+ hr = captureDevices->GetCount( &captureDeviceCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count.";
+ goto Exit;
+ }
+
+ // Count render devices
+ hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device collection.";
+ goto Exit;
+ }
+
+ hr = renderDevices->GetCount( &renderDeviceCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count.";
+ goto Exit;
+ }
+
+ // validate device index
+ if ( device >= captureDeviceCount + renderDeviceCount ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index.";
+ errorType = RtAudioError::INVALID_USE;
+ goto Exit;
+ }
+
+ // determine whether index falls within capture or render devices
+ if ( device >= renderDeviceCount ) {
+ hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device handle.";
+ goto Exit;
+ }
+ isCaptureDevice = true;
+ }
+ else {
+ hr = renderDevices->Item( device, &devicePtr );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device handle.";
+ goto Exit;
+ }
+ isCaptureDevice = false;
+ }
+
+ // get default device name
+ if ( isCaptureDevice ) {
+ hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default capture device handle.";
+ goto Exit;
+ }
+ }
+ else {
+ hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default render device handle.";
+ goto Exit;
+ }
+ }
+
+ hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store.";
+ goto Exit;
+ }
+ PropVariantInit( &defaultDeviceNameProp );
+
+ hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName.";
+ goto Exit;
+ }
+
+ defaultDeviceName = convertCharPointerToStdString(defaultDeviceNameProp.pwszVal);
+
+ // name
+ hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store.";
+ goto Exit;
+ }
+
+ PropVariantInit( &deviceNameProp );
+
+ hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName.";
+ goto Exit;
+ }
+
+ info.name =convertCharPointerToStdString(deviceNameProp.pwszVal);
+
+ // is default
+ if ( isCaptureDevice ) {
+ info.isDefaultInput = info.name == defaultDeviceName;
+ info.isDefaultOutput = false;
+ }
+ else {
+ info.isDefaultInput = false;
+ info.isDefaultOutput = info.name == defaultDeviceName;
+ }
+
+ // channel count
+ hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device audio client.";
+ goto Exit;
+ }
+
+ hr = audioClient->GetMixFormat( &deviceFormat );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format.";
+ goto Exit;
+ }
+
+ if ( isCaptureDevice ) {
+ info.inputChannels = deviceFormat->nChannels;
+ info.outputChannels = 0;
+ info.duplexChannels = 0;
+ }
+ else {
+ info.inputChannels = 0;
+ info.outputChannels = deviceFormat->nChannels;
+ info.duplexChannels = 0;
+ }
+
+ // sample rates
+ info.sampleRates.clear();
+
+ // allow support for all sample rates as we have a built-in sample rate converter
+ for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) {
+ info.sampleRates.push_back( SAMPLE_RATES[i] );
+ }
+ info.preferredSampleRate = deviceFormat->nSamplesPerSec;
+
+ // native format
+ info.nativeFormats = 0;
+
+ if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT ||
+ ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
+ ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) )
+ {
+ if ( deviceFormat->wBitsPerSample == 32 ) {
+ info.nativeFormats |= RTAUDIO_FLOAT32;
+ }
+ else if ( deviceFormat->wBitsPerSample == 64 ) {
+ info.nativeFormats |= RTAUDIO_FLOAT64;
+ }
+ }
+ else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM ||
+ ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
+ ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) )
+ {
+ if ( deviceFormat->wBitsPerSample == 8 ) {
+ info.nativeFormats |= RTAUDIO_SINT8;
+ }
+ else if ( deviceFormat->wBitsPerSample == 16 ) {
+ info.nativeFormats |= RTAUDIO_SINT16;
+ }
+ else if ( deviceFormat->wBitsPerSample == 24 ) {
+ info.nativeFormats |= RTAUDIO_SINT24;
+ }
+ else if ( deviceFormat->wBitsPerSample == 32 ) {
+ info.nativeFormats |= RTAUDIO_SINT32;
+ }
+ }
+
+ // probed
+ info.probed = true;
+
+Exit:
+ // release all references
+ PropVariantClear( &deviceNameProp );
+ PropVariantClear( &defaultDeviceNameProp );
+
+ SAFE_RELEASE( captureDevices );
+ SAFE_RELEASE( renderDevices );
+ SAFE_RELEASE( devicePtr );
+ SAFE_RELEASE( defaultDevicePtr );
+ SAFE_RELEASE( audioClient );
+ SAFE_RELEASE( devicePropStore );
+ SAFE_RELEASE( defaultDevicePropStore );
+
+ CoTaskMemFree( deviceFormat );
+ CoTaskMemFree( closestMatchFormat );
+
+ if ( !errorText_.empty() )
+ error( errorType );
+ return info;
+}
+
+//-----------------------------------------------------------------------------
+
+unsigned int RtApiWasapi::getDefaultOutputDevice( void )
+{
+ for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
+ if ( getDeviceInfo( i ).isDefaultOutput ) {
+ return i;
+ }
+ }
+
+ return 0;
+}
+
+//-----------------------------------------------------------------------------
+
+unsigned int RtApiWasapi::getDefaultInputDevice( void )
+{
+ for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
+ if ( getDeviceInfo( i ).isDefaultInput ) {
+ return i;
+ }
+ }
+
+ return 0;
+}
+
+//-----------------------------------------------------------------------------
+
+void RtApiWasapi::closeStream( void )
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiWasapi::closeStream: No open stream to close.";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ if ( stream_.state != STREAM_STOPPED )
+ stopStream();
+
+ // clean up stream memory
+ SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient )
+ SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient )
+
+ SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient )
+ SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient )
+
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent )
+ CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent );
+
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent )
+ CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent );
+
+ delete ( WasapiHandle* ) stream_.apiHandle;
+ stream_.apiHandle = NULL;
+
+ for ( int i = 0; i < 2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ // update stream state
+ stream_.state = STREAM_CLOSED;
+}
+
+//-----------------------------------------------------------------------------
+
+void RtApiWasapi::startStream( void )
+{
+ verifyStream();
+
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiWasapi::startStream: The stream is already running.";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ // update stream state
+ stream_.state = STREAM_RUNNING;
+
+ // create WASAPI stream thread
+ stream_.callbackInfo.thread = ( ThreadHandle ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL );
+
+ if ( !stream_.callbackInfo.thread ) {
+ errorText_ = "RtApiWasapi::startStream: Unable to instantiate callback thread.";
+ error( RtAudioError::THREAD_ERROR );
+ }
+ else {
+ SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority );
+ ResumeThread( ( void* ) stream_.callbackInfo.thread );
+ }
+}
+
+//-----------------------------------------------------------------------------
+
+void RtApiWasapi::stopStream( void )
+{
+ verifyStream();
+
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiWasapi::stopStream: The stream is already stopped.";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ // inform stream thread by setting stream state to STREAM_STOPPING
+ stream_.state = STREAM_STOPPING;
+
+ // wait until stream thread is stopped
+ while( stream_.state != STREAM_STOPPED ) {
+ Sleep( 1 );
+ }
+
+ // Wait for the last buffer to play before stopping.
+ Sleep( 1000 * stream_.bufferSize / stream_.sampleRate );
+
+ // stop capture client if applicable
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
+ HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream.";
+ error( RtAudioError::DRIVER_ERROR );
+ return;
+ }
+ }
+
+ // stop render client if applicable
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
+ HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream.";
+ error( RtAudioError::DRIVER_ERROR );
+ return;
+ }
+ }
+
+ // close thread handle
+ if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
+ errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread.";
+ error( RtAudioError::THREAD_ERROR );
+ return;
+ }
+
+ stream_.callbackInfo.thread = (ThreadHandle) NULL;
+}
+
+//-----------------------------------------------------------------------------
+
+void RtApiWasapi::abortStream( void )
+{
+ verifyStream();
+
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiWasapi::abortStream: The stream is already stopped.";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ // inform stream thread by setting stream state to STREAM_STOPPING
+ stream_.state = STREAM_STOPPING;
+
+ // wait until stream thread is stopped
+ while ( stream_.state != STREAM_STOPPED ) {
+ Sleep( 1 );
+ }
+
+ // stop capture client if applicable
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
+ HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::abortStream: Unable to stop capture stream.";
+ error( RtAudioError::DRIVER_ERROR );
+ return;
+ }
+ }
+
+ // stop render client if applicable
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
+ HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::abortStream: Unable to stop render stream.";
+ error( RtAudioError::DRIVER_ERROR );
+ return;
+ }
+ }
+
+ // close thread handle
+ if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
+ errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread.";
+ error( RtAudioError::THREAD_ERROR );
+ return;
+ }
+
+ stream_.callbackInfo.thread = (ThreadHandle) NULL;
+}
+
+//-----------------------------------------------------------------------------
+
+bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int* bufferSize,
+ RtAudio::StreamOptions* options )
+{
+ bool methodResult = FAILURE;
+ unsigned int captureDeviceCount = 0;
+ unsigned int renderDeviceCount = 0;
+
+ IMMDeviceCollection* captureDevices = NULL;
+ IMMDeviceCollection* renderDevices = NULL;
+ IMMDevice* devicePtr = NULL;
+ WAVEFORMATEX* deviceFormat = NULL;
+ unsigned int bufferBytes;
+ stream_.state = STREAM_STOPPED;
+
+ // create API Handle if not already created
+ if ( !stream_.apiHandle )
+ stream_.apiHandle = ( void* ) new WasapiHandle();
+
+ // Count capture devices
+ errorText_.clear();
+ RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
+ HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device collection.";
+ goto Exit;
+ }
+
+ hr = captureDevices->GetCount( &captureDeviceCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count.";
+ goto Exit;
+ }
+
+ // Count render devices
+ hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device collection.";
+ goto Exit;
+ }
+
+ hr = renderDevices->GetCount( &renderDeviceCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count.";
+ goto Exit;
+ }
+
+ // validate device index
+ if ( device >= captureDeviceCount + renderDeviceCount ) {
+ errorType = RtAudioError::INVALID_USE;
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index.";
+ goto Exit;
+ }
+
+ // determine whether index falls within capture or render devices
+ if ( device >= renderDeviceCount ) {
+ if ( mode != INPUT ) {
+ errorType = RtAudioError::INVALID_USE;
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device.";
+ goto Exit;
+ }
+
+ // retrieve captureAudioClient from devicePtr
+ IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
+
+ hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle.";
+ goto Exit;
+ }
+
+ hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
+ NULL, ( void** ) &captureAudioClient );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
+ goto Exit;
+ }
+
+ hr = captureAudioClient->GetMixFormat( &deviceFormat );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
+ goto Exit;
+ }
+
+ stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
+ captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
+ }
+ else {
+ if ( mode != OUTPUT ) {
+ errorType = RtAudioError::INVALID_USE;
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Render device selected as input device.";
+ goto Exit;
+ }
+
+ // retrieve renderAudioClient from devicePtr
+ IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
+
+ hr = renderDevices->Item( device, &devicePtr );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
+ goto Exit;
+ }
+
+ hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
+ NULL, ( void** ) &renderAudioClient );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
+ goto Exit;
+ }
+
+ hr = renderAudioClient->GetMixFormat( &deviceFormat );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
+ goto Exit;
+ }
+
+ stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
+ renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
+ }
+
+ // fill stream data
+ if ( ( stream_.mode == OUTPUT && mode == INPUT ) ||
+ ( stream_.mode == INPUT && mode == OUTPUT ) ) {
+ stream_.mode = DUPLEX;
+ }
+ else {
+ stream_.mode = mode;
+ }
+
+ stream_.device[mode] = device;
+ stream_.doByteSwap[mode] = false;
+ stream_.sampleRate = sampleRate;
+ stream_.bufferSize = *bufferSize;
+ stream_.nBuffers = 1;
+ stream_.nUserChannels[mode] = channels;
+ stream_.channelOffset[mode] = firstChannel;
+ stream_.userFormat = format;
+ stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats;
+
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
+ stream_.userInterleaved = false;
+ else
+ stream_.userInterleaved = true;
+ stream_.deviceInterleaved[mode] = true;
+
+ // Set flags for buffer conversion.
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] ||
+ stream_.nUserChannels != stream_.nDeviceChannels )
+ stream_.doConvertBuffer[mode] = true;
+ else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
+
+ if ( stream_.doConvertBuffer[mode] )
+ setConvertInfo( mode, 0 );
+
+ // Allocate necessary internal buffers
+ bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat );
+
+ stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 );
+ if ( !stream_.userBuffer[mode] ) {
+ errorType = RtAudioError::MEMORY_ERROR;
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory.";
+ goto Exit;
+ }
+
+ if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME )
+ stream_.callbackInfo.priority = 15;
+ else
+ stream_.callbackInfo.priority = 0;
+
+ ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback
+ ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode
+
+ methodResult = SUCCESS;
+
+Exit:
+ //clean up
+ SAFE_RELEASE( captureDevices );
+ SAFE_RELEASE( renderDevices );
+ SAFE_RELEASE( devicePtr );
+ CoTaskMemFree( deviceFormat );
+
+ // if method failed, close the stream
+ if ( methodResult == FAILURE )
+ closeStream();
+
+ if ( !errorText_.empty() )
+ error( errorType );
+ return methodResult;
+}
+
+//=============================================================================
+
+DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr )
+{
+ if ( wasapiPtr )
+ ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread();
+
+ return 0;
+}
+
+DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr )
+{
+ if ( wasapiPtr )
+ ( ( RtApiWasapi* ) wasapiPtr )->stopStream();
+
+ return 0;
+}
+
+DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr )
+{
+ if ( wasapiPtr )
+ ( ( RtApiWasapi* ) wasapiPtr )->abortStream();
+
+ return 0;
+}
+
+//-----------------------------------------------------------------------------
+
+void RtApiWasapi::wasapiThread()
+{
+ // as this is a new thread, we must CoInitialize it
+ CoInitialize( NULL );
+
+ HRESULT hr;
+
+ IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
+ IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
+ IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient;
+ IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient;
+ HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent;
+ HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent;
+
+ WAVEFORMATEX* captureFormat = NULL;
+ WAVEFORMATEX* renderFormat = NULL;
+ float captureSrRatio = 0.0f;
+ float renderSrRatio = 0.0f;
+ WasapiBuffer captureBuffer;
+ WasapiBuffer renderBuffer;
+
+ // declare local stream variables
+ RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;
+ BYTE* streamBuffer = NULL;
+ unsigned long captureFlags = 0;
+ unsigned int bufferFrameCount = 0;
+ unsigned int numFramesPadding = 0;
+ unsigned int convBufferSize = 0;
+ bool callbackPushed = false;
+ bool callbackPulled = false;
+ bool callbackStopped = false;
+ int callbackResult = 0;
+
+ // convBuffer is used to store converted buffers between WASAPI and the user
+ char* convBuffer = NULL;
+ unsigned int convBuffSize = 0;
+ unsigned int deviceBuffSize = 0;
+
+ errorText_.clear();
+ RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
+
+ // Attempt to assign "Pro Audio" characteristic to thread
+ HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" );
+ if ( AvrtDll ) {
+ DWORD taskIndex = 0;
+ TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );
+ AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex );
+ FreeLibrary( AvrtDll );
+ }
+
+ // start capture stream if applicable
+ if ( captureAudioClient ) {
+ hr = captureAudioClient->GetMixFormat( &captureFormat );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
+ goto Exit;
+ }
+
+ captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate );
+
+ // initialize capture stream according to desire buffer size
+ float desiredBufferSize = stream_.bufferSize * captureSrRatio;
+ REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / captureFormat->nSamplesPerSec );
+
+ if ( !captureClient ) {
+ hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
+ AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
+ desiredBufferPeriod,
+ desiredBufferPeriod,
+ captureFormat,
+ NULL );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client.";
+ goto Exit;
+ }
+
+ hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ),
+ ( void** ) &captureClient );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle.";
+ goto Exit;
+ }
+
+ // configure captureEvent to trigger on every available capture buffer
+ captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
+ if ( !captureEvent ) {
+ errorType = RtAudioError::SYSTEM_ERROR;
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to create capture event.";
+ goto Exit;
+ }
+
+ hr = captureAudioClient->SetEventHandle( captureEvent );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to set capture event handle.";
+ goto Exit;
+ }
+
+ ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;
+ ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent;
+ }
+
+ unsigned int inBufferSize = 0;
+ hr = captureAudioClient->GetBufferSize( &inBufferSize );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to get capture buffer size.";
+ goto Exit;
+ }
+
+ // scale outBufferSize according to stream->user sample rate ratio
+ unsigned int outBufferSize = ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT];
+ inBufferSize *= stream_.nDeviceChannels[INPUT];
+
+ // set captureBuffer size
+ captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) );
+
+ // reset the capture stream
+ hr = captureAudioClient->Reset();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to reset capture stream.";
+ goto Exit;
+ }
+
+ // start the capture stream
+ hr = captureAudioClient->Start();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to start capture stream.";
+ goto Exit;
+ }
+ }
+
+ // start render stream if applicable
+ if ( renderAudioClient ) {
+ hr = renderAudioClient->GetMixFormat( &renderFormat );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
+ goto Exit;
+ }
+
+ renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate );
+
+ // initialize render stream according to desire buffer size
+ float desiredBufferSize = stream_.bufferSize * renderSrRatio;
+ REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / renderFormat->nSamplesPerSec );
+
+ if ( !renderClient ) {
+ hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
+ AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
+ desiredBufferPeriod,
+ desiredBufferPeriod,
+ renderFormat,
+ NULL );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize render audio client.";
+ goto Exit;
+ }
+
+ hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ),
+ ( void** ) &renderClient );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle.";
+ goto Exit;
+ }
+
+ // configure renderEvent to trigger on every available render buffer
+ renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
+ if ( !renderEvent ) {
+ errorType = RtAudioError::SYSTEM_ERROR;
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to create render event.";
+ goto Exit;
+ }
+
+ hr = renderAudioClient->SetEventHandle( renderEvent );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to set render event handle.";
+ goto Exit;
+ }
+
+ ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient;
+ ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent;
+ }
+
+ unsigned int outBufferSize = 0;
+ hr = renderAudioClient->GetBufferSize( &outBufferSize );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to get render buffer size.";
+ goto Exit;
+ }
+
+ // scale inBufferSize according to user->stream sample rate ratio
+ unsigned int inBufferSize = ( unsigned int ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT];
+ outBufferSize *= stream_.nDeviceChannels[OUTPUT];
+
+ // set renderBuffer size
+ renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) );
+
+ // reset the render stream
+ hr = renderAudioClient->Reset();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to reset render stream.";
+ goto Exit;
+ }
+
+ // start the render stream
+ hr = renderAudioClient->Start();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to start render stream.";
+ goto Exit;
+ }
+ }
+
+ if ( stream_.mode == INPUT ) {
+ convBuffSize = ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
+ deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
+ }
+ else if ( stream_.mode == OUTPUT ) {
+ convBuffSize = ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
+ deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
+ }
+ else if ( stream_.mode == DUPLEX ) {
+ convBuffSize = std::max( ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
+ ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
+ deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
+ stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
+ }
+
+ convBuffer = ( char* ) malloc( convBuffSize );
+ stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize );
+ if ( !convBuffer || !stream_.deviceBuffer ) {
+ errorType = RtAudioError::MEMORY_ERROR;
+ errorText_ = "RtApiWasapi::wasapiThread: Error allocating device buffer memory.";
+ goto Exit;
+ }
+
+ // stream process loop
+ while ( stream_.state != STREAM_STOPPING ) {
+ if ( !callbackPulled ) {
+ // Callback Input
+ // ==============
+ // 1. Pull callback buffer from inputBuffer
+ // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count
+ // Convert callback buffer to user format
+
+ if ( captureAudioClient ) {
+ // Pull callback buffer from inputBuffer
+ callbackPulled = captureBuffer.pullBuffer( convBuffer,
+ ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT],
+ stream_.deviceFormat[INPUT] );
+
+ if ( callbackPulled ) {
+ // Convert callback buffer to user sample rate
+ convertBufferWasapi( stream_.deviceBuffer,
+ convBuffer,
+ stream_.nDeviceChannels[INPUT],
+ captureFormat->nSamplesPerSec,
+ stream_.sampleRate,
+ ( unsigned int ) ( stream_.bufferSize * captureSrRatio ),
+ convBufferSize,
+ stream_.deviceFormat[INPUT] );
+
+ if ( stream_.doConvertBuffer[INPUT] ) {
+ // Convert callback buffer to user format
+ convertBuffer( stream_.userBuffer[INPUT],
+ stream_.deviceBuffer,
+ stream_.convertInfo[INPUT] );
+ }
+ else {
+ // no further conversion, simple copy deviceBuffer to userBuffer
+ memcpy( stream_.userBuffer[INPUT],
+ stream_.deviceBuffer,
+ stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) );
+ }
+ }
+ }
+ else {
+ // if there is no capture stream, set callbackPulled flag
+ callbackPulled = true;
+ }
+
+ // Execute Callback
+ // ================
+ // 1. Execute user callback method
+ // 2. Handle return value from callback
+
+ // if callback has not requested the stream to stop
+ if ( callbackPulled && !callbackStopped ) {
+ // Execute user callback method
+ callbackResult = callback( stream_.userBuffer[OUTPUT],
+ stream_.userBuffer[INPUT],
+ stream_.bufferSize,
+ getStreamTime(),
+ captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,
+ stream_.callbackInfo.userData );
+
+ // Handle return value from callback
+ if ( callbackResult == 1 ) {
+ // instantiate a thread to stop this thread
+ HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL );
+ if ( !threadHandle ) {
+ errorType = RtAudioError::THREAD_ERROR;
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread.";
+ goto Exit;
+ }
+ else if ( !CloseHandle( threadHandle ) ) {
+ errorType = RtAudioError::THREAD_ERROR;
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle.";
+ goto Exit;
+ }
+
+ callbackStopped = true;
+ }
+ else if ( callbackResult == 2 ) {
+ // instantiate a thread to stop this thread
+ HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL );
+ if ( !threadHandle ) {
+ errorType = RtAudioError::THREAD_ERROR;
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread.";
+ goto Exit;
+ }
+ else if ( !CloseHandle( threadHandle ) ) {
+ errorType = RtAudioError::THREAD_ERROR;
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle.";
+ goto Exit;
+ }
+
+ callbackStopped = true;
+ }
+ }
+ }
+
+ // Callback Output
+ // ===============
+ // 1. Convert callback buffer to stream format
+ // 2. Convert callback buffer to stream sample rate and channel count
+ // 3. Push callback buffer into outputBuffer
+
+ if ( renderAudioClient && callbackPulled ) {
+ if ( stream_.doConvertBuffer[OUTPUT] ) {
+ // Convert callback buffer to stream format
+ convertBuffer( stream_.deviceBuffer,
+ stream_.userBuffer[OUTPUT],
+ stream_.convertInfo[OUTPUT] );
+
+ }
+
+ // Convert callback buffer to stream sample rate
+ convertBufferWasapi( convBuffer,
+ stream_.deviceBuffer,
+ stream_.nDeviceChannels[OUTPUT],
+ stream_.sampleRate,
+ renderFormat->nSamplesPerSec,
+ stream_.bufferSize,
+ convBufferSize,
+ stream_.deviceFormat[OUTPUT] );
+
+ // Push callback buffer into outputBuffer
+ callbackPushed = renderBuffer.pushBuffer( convBuffer,
+ convBufferSize * stream_.nDeviceChannels[OUTPUT],
+ stream_.deviceFormat[OUTPUT] );
+ }
+ else {
+ // if there is no render stream, set callbackPushed flag
+ callbackPushed = true;
+ }
+
+ // Stream Capture
+ // ==============
+ // 1. Get capture buffer from stream
+ // 2. Push capture buffer into inputBuffer
+ // 3. If 2. was successful: Release capture buffer
+
+ if ( captureAudioClient ) {
+ // if the callback input buffer was not pulled from captureBuffer, wait for next capture event
+ if ( !callbackPulled ) {
+ WaitForSingleObject( captureEvent, INFINITE );
+ }
+
+ // Get capture buffer from stream
+ hr = captureClient->GetBuffer( &streamBuffer,
+ &bufferFrameCount,
+ &captureFlags, NULL, NULL );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer.";
+ goto Exit;
+ }
+
+ if ( bufferFrameCount != 0 ) {
+ // Push capture buffer into inputBuffer
+ if ( captureBuffer.pushBuffer( ( char* ) streamBuffer,
+ bufferFrameCount * stream_.nDeviceChannels[INPUT],
+ stream_.deviceFormat[INPUT] ) )
+ {
+ // Release capture buffer
+ hr = captureClient->ReleaseBuffer( bufferFrameCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
+ goto Exit;
+ }
+ }
+ else
+ {
+ // Inform WASAPI that capture was unsuccessful
+ hr = captureClient->ReleaseBuffer( 0 );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
+ goto Exit;
+ }
+ }
+ }
+ else
+ {
+ // Inform WASAPI that capture was unsuccessful
+ hr = captureClient->ReleaseBuffer( 0 );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
+ goto Exit;
+ }
+ }
+ }
+
+ // Stream Render
+ // =============
+ // 1. Get render buffer from stream
+ // 2. Pull next buffer from outputBuffer
+ // 3. If 2. was successful: Fill render buffer with next buffer
+ // Release render buffer
+
+ if ( renderAudioClient ) {
+ // if the callback output buffer was not pushed to renderBuffer, wait for next render event
+ if ( callbackPulled && !callbackPushed ) {
+ WaitForSingleObject( renderEvent, INFINITE );
+ }
+
+ // Get render buffer from stream
+ hr = renderAudioClient->GetBufferSize( &bufferFrameCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size.";
+ goto Exit;
+ }
+
+ hr = renderAudioClient->GetCurrentPadding( &numFramesPadding );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding.";
+ goto Exit;
+ }
+
+ bufferFrameCount -= numFramesPadding;
+
+ if ( bufferFrameCount != 0 ) {
+ hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer.";
+ goto Exit;
+ }
+
+ // Pull next buffer from outputBuffer
+ // Fill render buffer with next buffer
+ if ( renderBuffer.pullBuffer( ( char* ) streamBuffer,
+ bufferFrameCount * stream_.nDeviceChannels[OUTPUT],
+ stream_.deviceFormat[OUTPUT] ) )
+ {
+ // Release render buffer
+ hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
+ goto Exit;
+ }
+ }
+ else
+ {
+ // Inform WASAPI that render was unsuccessful
+ hr = renderClient->ReleaseBuffer( 0, 0 );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
+ goto Exit;
+ }
+ }
+ }
+ else
+ {
+ // Inform WASAPI that render was unsuccessful
+ hr = renderClient->ReleaseBuffer( 0, 0 );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
+ goto Exit;
+ }
+ }
+ }
+
+ // if the callback buffer was pushed renderBuffer reset callbackPulled flag
+ if ( callbackPushed ) {
+ callbackPulled = false;
+ // tick stream time
+ RtApi::tickStreamTime();
+ }
+
+ }
+
+Exit:
+ // clean up
+ CoTaskMemFree( captureFormat );
+ CoTaskMemFree( renderFormat );
+
+ free ( convBuffer );
+
+ CoUninitialize();
+
+ // update stream state
+ stream_.state = STREAM_STOPPED;
+
+ if ( errorText_.empty() )
+ return;
+ else
+ error( errorType );
+}
+
+//******************** End of __WINDOWS_WASAPI__ *********************//
+#endif
+
+
+#if defined(__WINDOWS_DS__) // Windows DirectSound API
+
+// Modified by Robin Davies, October 2005
+// - Improvements to DirectX pointer chasing.
+// - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.
+// - Auto-call CoInitialize for DSOUND and ASIO platforms.
+// Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
+// Changed device query structure for RtAudio 4.0.7, January 2010
+
+#include <dsound.h>
+#include <assert.h>
+#include <algorithm>
+
+#if defined(__MINGW32__)
+ // missing from latest mingw winapi
+#define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */
+#define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */
+#define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */
+#define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */
+#endif
+
+#define MINIMUM_DEVICE_BUFFER_SIZE 32768
+
+#ifdef _MSC_VER // if Microsoft Visual C++
+#pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually.
+#endif
+
+static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
+{
+ if ( pointer > bufferSize ) pointer -= bufferSize;
+ if ( laterPointer < earlierPointer ) laterPointer += bufferSize;
+ if ( pointer < earlierPointer ) pointer += bufferSize;
+ return pointer >= earlierPointer && pointer < laterPointer;
+}
+
+// A structure to hold various information related to the DirectSound
+// API implementation.
+struct DsHandle {
+ unsigned int drainCounter; // Tracks callback counts when draining
+ bool internalDrain; // Indicates if stop is initiated from callback or not.
+ void *id[2];
+ void *buffer[2];
+ bool xrun[2];
+ UINT bufferPointer[2];
+ DWORD dsBufferSize[2];
+ DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.
+ HANDLE condition;
+
+ DsHandle()
+ :drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; }
+};
+
+// 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( int code );
+
+static unsigned __stdcall callbackHandler( void *ptr );
+
+struct DsDevice {
+ LPGUID id[2];
+ bool validId[2];
+ bool found;
+ std::string name;
+
+ DsDevice()
+ : found(false) { validId[0] = false; validId[1] = false; }
+};
+
+struct DsProbeData {
+ bool isInput;
+ std::vector<struct DsDevice>* dsDevices;
+};
+
+RtApiDs :: RtApiDs()
+{
+ // Dsound will run both-threaded. If CoInitialize fails, then just
+ // accept whatever the mainline chose for a threading model.
+ coInitialized_ = false;
+ HRESULT hr = CoInitialize( NULL );
+ if ( !FAILED( hr ) ) coInitialized_ = true;
+}
+
+RtApiDs :: ~RtApiDs()
+{
+ if ( coInitialized_ ) CoUninitialize(); // balanced call.
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
+}
+
+// The DirectSound default output is always the first device.
+unsigned int RtApiDs :: getDefaultOutputDevice( void )
+{
+ return 0;
+}
+
+// The DirectSound default input is always the first input device,
+// which is the first capture device enumerated.
+unsigned int RtApiDs :: getDefaultInputDevice( void )
+{
+ return 0;
+}
+
+unsigned int RtApiDs :: getDeviceCount( void )
+{
+ // Set query flag for previously found devices to false, so that we
+ // can check for any devices that have disappeared.
+ for ( unsigned int i=0; i<dsDevices.size(); i++ )
+ dsDevices[i].found = false;
+
+ // Query DirectSound devices.
+ struct DsProbeData probeInfo;
+ probeInfo.isInput = false;
+ probeInfo.dsDevices = &dsDevices;
+ HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ }
+
+ // Query DirectSoundCapture devices.
+ probeInfo.isInput = true;
+ result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ }
+
+ // Clean out any devices that may have disappeared (code update submitted by Eli Zehngut).
+ for ( unsigned int i=0; i<dsDevices.size(); ) {
+ if ( dsDevices[i].found == false ) dsDevices.erase( dsDevices.begin() + i );
+ else i++;
+ }
+
+ return static_cast<unsigned int>(dsDevices.size());
+}
+
+RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
+{
+ RtAudio::DeviceInfo info;
+ info.probed = false;
+
+ if ( dsDevices.size() == 0 ) {
+ // Force a query of all devices
+ getDeviceCount();
+ if ( dsDevices.size() == 0 ) {
+ errorText_ = "RtApiDs::getDeviceInfo: no devices found!";
+ error( RtAudioError::INVALID_USE );
+ return info;
+ }
+ }
+
+ if ( device >= dsDevices.size() ) {
+ errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!";
+ error( RtAudioError::INVALID_USE );
+ return info;
+ }
+
+ HRESULT result;
+ if ( dsDevices[ device ].validId[0] == false ) goto probeInput;
+
+ LPDIRECTSOUND output;
+ DSCAPS outCaps;
+ result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto probeInput;
+ }
+
+ outCaps.dwSize = sizeof( outCaps );
+ result = output->GetCaps( &outCaps );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto probeInput;
+ }
+
+ // Get output channel information.
+ info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
+
+ // Get sample rate information.
+ info.sampleRates.clear();
+ for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
+ if ( SAMPLE_RATES[k] >= (unsigned int) outCaps.dwMinSecondarySampleRate &&
+ SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) {
+ info.sampleRates.push_back( SAMPLE_RATES[k] );
+
+ if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
+ info.preferredSampleRate = SAMPLE_RATES[k];
+ }
+ }
+
+ // Get format information.
+ if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8;
+
+ output->Release();
+
+ if ( getDefaultOutputDevice() == device )
+ info.isDefaultOutput = true;
+
+ if ( dsDevices[ device ].validId[1] == false ) {
+ info.name = dsDevices[ device ].name;
+ info.probed = true;
+ return info;
+ }
+
+ probeInput:
+
+ LPDIRECTSOUNDCAPTURE input;
+ result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ DSCCAPS inCaps;
+ inCaps.dwSize = sizeof( inCaps );
+ result = input->GetCaps( &inCaps );
+ if ( FAILED( result ) ) {
+ input->Release();
+ errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // Get input channel information.
+ info.inputChannels = inCaps.dwChannels;
+
+ // Get sample rate and format information.
+ std::vector<unsigned int> rates;
+ if ( inCaps.dwChannels >= 2 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8;
+
+ if ( info.nativeFormats & RTAUDIO_SINT16 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 );
+ }
+ else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 );
+ }
+ }
+ else if ( inCaps.dwChannels == 1 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8;
+
+ if ( info.nativeFormats & RTAUDIO_SINT16 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 );
+ }
+ else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 );
+ }
+ }
+ else info.inputChannels = 0; // technically, this would be an error
+
+ input->Release();
+
+ if ( info.inputChannels == 0 ) return info;
+
+ // Copy the supported rates to the info structure but avoid duplication.
+ bool found;
+ for ( unsigned int i=0; i<rates.size(); i++ ) {
+ found = false;
+ for ( unsigned int 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() );
+
+ // 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 = dsDevices[ device ].name;
+ info.probed = true;
+ return info;
+}
+
+bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options )
+{
+ if ( channels + firstChannel > 2 ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";
+ return FAILURE;
+ }
+
+ size_t nDevices = dsDevices.size();
+ if ( nDevices == 0 ) {
+ // This should not happen because a check is made before this function is called.
+ errorText_ = "RtApiDs::probeDeviceOpen: no devices found!";
+ return FAILURE;
+ }
+
+ if ( device >= nDevices ) {
+ // This should not happen because a check is made before this function is called.
+ errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!";
+ return FAILURE;
+ }
+
+ if ( mode == OUTPUT ) {
+ if ( dsDevices[ device ].validId[0] == false ) {
+ errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+ else { // mode == INPUT
+ if ( dsDevices[ device ].validId[1] == false ) {
+ errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+
+ // 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
+ // window. Also, if the application window closes before the
+ // DirectSound buffer, DirectSound can crash. In the past, I had
+ // problems when using GetDesktopWindow() but it seems fine now
+ // (January 2010). I'll leave it commented here.
+ // HWND hWnd = GetForegroundWindow();
+ HWND hWnd = GetDesktopWindow();
+
+ // Check the numberOfBuffers parameter and limit the lowest value to
+ // two. This is a judgement call and a value of two is probably too
+ // low for capture, but it should work for playback.
+ int nBuffers = 0;
+ if ( options ) nBuffers = options->numberOfBuffers;
+ if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2;
+ if ( nBuffers < 2 ) nBuffers = 3;
+
+ // Check the lower range of the user-specified buffer size and set
+ // (arbitrarily) to a lower bound of 32.
+ if ( *bufferSize < 32 ) *bufferSize = 32;
+
+ // Create the wave format structure. The data format setting will
+ // be determined later.
+ WAVEFORMATEX waveFormat;
+ ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) );
+ waveFormat.wFormatTag = WAVE_FORMAT_PCM;
+ waveFormat.nChannels = channels + firstChannel;
+ waveFormat.nSamplesPerSec = (unsigned long) sampleRate;
+
+ // Determine the device buffer size. By default, we'll use the value
+ // defined above (32K), but we will grow it to make allowances for
+ // very large software buffer sizes.
+ DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE;
+ DWORD dsPointerLeadTime = 0;
+
+ void *ohandle = 0, *bhandle = 0;
+ HRESULT result;
+ if ( mode == OUTPUT ) {
+
+ LPDIRECTSOUND output;
+ result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ DSCAPS outCaps;
+ outCaps.dwSize = sizeof( outCaps );
+ result = output->GetCaps( &outCaps );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Check channel information.
+ if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {
+ errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Check format information. Use 16-bit format unless not
+ // supported or user requests 8-bit.
+ if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT &&
+ !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) {
+ waveFormat.wBitsPerSample = 16;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ }
+ else {
+ waveFormat.wBitsPerSample = 8;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+ }
+ stream_.userFormat = format;
+
+ // Update wave format structure and buffer information.
+ waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
+ waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
+ dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
+
+ // If the user wants an even bigger buffer, increase the device buffer size accordingly.
+ while ( dsPointerLeadTime * 2U > dsBufferSize )
+ dsBufferSize *= 2;
+
+ // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.
+ // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );
+ // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.
+ result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Even though we will write to the secondary buffer, we need to
+ // access the primary buffer to set the correct output format
+ // (since the default is 8-bit, 22 kHz!). Setup the DS primary
+ // buffer description.
+ DSBUFFERDESC bufferDescription;
+ ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
+ bufferDescription.dwSize = sizeof( DSBUFFERDESC );
+ bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
+
+ // Obtain the primary buffer
+ LPDIRECTSOUNDBUFFER buffer;
+ result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Set the primary DS buffer sound format.
+ result = buffer->SetFormat( &waveFormat );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Setup the secondary DS buffer description.
+ ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
+ bufferDescription.dwSize = sizeof( DSBUFFERDESC );
+ bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
+ DSBCAPS_GLOBALFOCUS |
+ DSBCAPS_GETCURRENTPOSITION2 |
+ DSBCAPS_LOCHARDWARE ); // Force hardware mixing
+ bufferDescription.dwBufferBytes = dsBufferSize;
+ bufferDescription.lpwfxFormat = &waveFormat;
+
+ // Try to create the secondary DS buffer. If that doesn't work,
+ // try to use software mixing. Otherwise, there's a problem.
+ result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
+ if ( FAILED( result ) ) {
+ bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
+ DSBCAPS_GLOBALFOCUS |
+ DSBCAPS_GETCURRENTPOSITION2 |
+ DSBCAPS_LOCSOFTWARE ); // Force software mixing
+ result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+
+ // Get the buffer size ... might be different from what we specified.
+ DSBCAPS dsbcaps;
+ dsbcaps.dwSize = sizeof( DSBCAPS );
+ result = buffer->GetCaps( &dsbcaps );
+ if ( FAILED( result ) ) {
+ output->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ dsBufferSize = dsbcaps.dwBufferBytes;
+
+ // Lock the DS buffer
+ LPVOID audioPtr;
+ DWORD dataLen;
+ result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
+ if ( FAILED( result ) ) {
+ output->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Zero the DS buffer
+ ZeroMemory( audioPtr, dataLen );
+
+ // Unlock the DS buffer
+ result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+ if ( FAILED( result ) ) {
+ output->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ ohandle = (void *) output;
+ bhandle = (void *) buffer;
+ }
+
+ if ( mode == INPUT ) {
+
+ LPDIRECTSOUNDCAPTURE input;
+ result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ DSCCAPS inCaps;
+ inCaps.dwSize = sizeof( inCaps );
+ result = input->GetCaps( &inCaps );
+ if ( FAILED( result ) ) {
+ input->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Check channel information.
+ if ( inCaps.dwChannels < channels + firstChannel ) {
+ errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";
+ return FAILURE;
+ }
+
+ // Check format information. Use 16-bit format unless user
+ // requests 8-bit.
+ DWORD deviceFormats;
+ if ( channels + firstChannel == 2 ) {
+ deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08;
+ if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
+ waveFormat.wBitsPerSample = 8;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+ }
+ else { // assume 16-bit is supported
+ waveFormat.wBitsPerSample = 16;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ }
+ }
+ else { // channel == 1
+ deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08;
+ if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
+ waveFormat.wBitsPerSample = 8;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+ }
+ else { // assume 16-bit is supported
+ waveFormat.wBitsPerSample = 16;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ }
+ }
+ stream_.userFormat = format;
+
+ // Update wave format structure and buffer information.
+ waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
+ waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
+ dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
+
+ // If the user wants an even bigger buffer, increase the device buffer size accordingly.
+ while ( dsPointerLeadTime * 2U > dsBufferSize )
+ dsBufferSize *= 2;
+
+ // Setup the secondary DS buffer description.
+ DSCBUFFERDESC bufferDescription;
+ ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );
+ bufferDescription.dwSize = sizeof( DSCBUFFERDESC );
+ bufferDescription.dwFlags = 0;
+ bufferDescription.dwReserved = 0;
+ bufferDescription.dwBufferBytes = dsBufferSize;
+ bufferDescription.lpwfxFormat = &waveFormat;
+
+ // Create the capture buffer.
+ LPDIRECTSOUNDCAPTUREBUFFER buffer;
+ result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );
+ if ( FAILED( result ) ) {
+ input->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Get the buffer size ... might be different from what we specified.
+ DSCBCAPS dscbcaps;
+ dscbcaps.dwSize = sizeof( DSCBCAPS );
+ result = buffer->GetCaps( &dscbcaps );
+ if ( FAILED( result ) ) {
+ input->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ dsBufferSize = dscbcaps.dwBufferBytes;
+
+ // NOTE: We could have a problem here if this is a duplex stream
+ // and the play and capture hardware buffer sizes are different
+ // (I'm actually not sure if that is a problem or not).
+ // Currently, we are not verifying that.
+
+ // Lock the capture buffer
+ LPVOID audioPtr;
+ DWORD dataLen;
+ result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
+ if ( FAILED( result ) ) {
+ input->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Zero the buffer
+ ZeroMemory( audioPtr, dataLen );
+
+ // Unlock the buffer
+ result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+ if ( FAILED( result ) ) {
+ input->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ ohandle = (void *) input;
+ bhandle = (void *) buffer;
+ }
+
+ // Set various stream parameters
+ DsHandle *handle = 0;
+ stream_.nDeviceChannels[mode] = channels + firstChannel;
+ stream_.nUserChannels[mode] = channels;
+ stream_.bufferSize = *bufferSize;
+ stream_.channelOffset[mode] = firstChannel;
+ stream_.deviceInterleaved[mode] = true;
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+ else stream_.userInterleaved = true;
+
+ // Set flag for buffer conversion
+ stream_.doConvertBuffer[mode] = false;
+ if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])
+ stream_.doConvertBuffer[mode] = true;
+ if (stream_.userFormat != stream_.deviceFormat[mode])
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
+
+ // Allocate necessary internal buffers
+ long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
+ }
+
+ if ( stream_.doConvertBuffer[mode] ) {
+
+ bool makeBuffer = true;
+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+ if ( mode == INPUT ) {
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ if ( bufferBytes <= (long) bytesOut ) makeBuffer = false;
+ }
+ }
+
+ if ( makeBuffer ) {
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory.";
+ goto error;
+ }
+ }
+ }
+
+ // Allocate our DsHandle structures for the stream.
+ if ( stream_.apiHandle == 0 ) {
+ try {
+ handle = new DsHandle;
+ }
+ catch ( std::bad_alloc& ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory.";
+ goto error;
+ }
+
+ // Create a manual-reset event.
+ handle->condition = CreateEvent( NULL, // no security
+ TRUE, // manual-reset
+ FALSE, // non-signaled initially
+ NULL ); // unnamed
+ stream_.apiHandle = (void *) handle;
+ }
+ else
+ handle = (DsHandle *) stream_.apiHandle;
+ handle->id[mode] = ohandle;
+ handle->buffer[mode] = bhandle;
+ handle->dsBufferSize[mode] = dsBufferSize;
+ handle->dsPointerLeadTime[mode] = dsPointerLeadTime;
+
+ stream_.device[mode] = device;
+ stream_.state = STREAM_STOPPED;
+ if ( stream_.mode == OUTPUT && mode == INPUT )
+ // We had already set up an output stream.
+ stream_.mode = DUPLEX;
+ else
+ stream_.mode = mode;
+ stream_.nBuffers = nBuffers;
+ stream_.sampleRate = sampleRate;
+
+ // Setup the buffer conversion information structure.
+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+
+ // Setup the callback thread.
+ if ( stream_.callbackInfo.isRunning == false ) {
+ unsigned threadId;
+ stream_.callbackInfo.isRunning = true;
+ stream_.callbackInfo.object = (void *) this;
+ stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,
+ &stream_.callbackInfo, 0, &threadId );
+ if ( stream_.callbackInfo.thread == 0 ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";
+ goto error;
+ }
+
+ // Boost DS thread priority
+ SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );
+ }
+ return SUCCESS;
+
+ error:
+ if ( handle ) {
+ if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
+ LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
+ LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ if ( buffer ) buffer->Release();
+ object->Release();
+ }
+ if ( handle->buffer[1] ) {
+ LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
+ LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ if ( buffer ) buffer->Release();
+ object->Release();
+ }
+ CloseHandle( handle->condition );
+ delete handle;
+ stream_.apiHandle = 0;
+ }
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ stream_.state = STREAM_CLOSED;
+ return FAILURE;
+}
+
+void RtApiDs :: closeStream()
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiDs::closeStream(): no open stream to close!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ // Stop the callback thread.
+ stream_.callbackInfo.isRunning = false;
+ WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );
+ CloseHandle( (HANDLE) stream_.callbackInfo.thread );
+
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
+ if ( handle ) {
+ if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
+ LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
+ LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ if ( buffer ) {
+ buffer->Stop();
+ buffer->Release();
+ }
+ object->Release();
+ }
+ if ( handle->buffer[1] ) {
+ LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
+ LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ if ( buffer ) {
+ buffer->Stop();
+ buffer->Release();
+ }
+ object->Release();
+ }
+ CloseHandle( handle->condition );
+ delete handle;
+ stream_.apiHandle = 0;
+ }
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
+}
+
+void RtApiDs :: startStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiDs::startStream(): the stream is already running!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
+
+ // Increase scheduler frequency on lesser windows (a side-effect of
+ // increasing timer accuracy). On greater windows (Win2K or later),
+ // this is already in effect.
+ timeBeginPeriod( 1 );
+
+ buffersRolling = false;
+ duplexPrerollBytes = 0;
+
+ if ( stream_.mode == DUPLEX ) {
+ // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize.
+ duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );
+ }
+
+ HRESULT result = 0;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+ LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ result = buffer->Play( 0, 0, DSBPLAY_LOOPING );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+
+ LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ result = buffer->Start( DSCBSTART_LOOPING );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ handle->drainCounter = 0;
+ handle->internalDrain = false;
+ ResetEvent( handle->condition );
+ stream_.state = STREAM_RUNNING;
+
+ unlock:
+ if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiDs :: stopStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ HRESULT result = 0;
+ LPVOID audioPtr;
+ DWORD dataLen;
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ if ( handle->drainCounter == 0 ) {
+ handle->drainCounter = 2;
+ WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
+ }
+
+ stream_.state = STREAM_STOPPED;
+
+ MUTEX_LOCK( &stream_.mutex );
+
+ // Stop the buffer and clear memory
+ LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ result = buffer->Stop();
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!";
+ errorText_ = errorStream_.str();
+ 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, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+
+ // Zero the DS buffer
+ ZeroMemory( audioPtr, dataLen );
+
+ // Unlock the DS buffer
+ result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+
+ // If we start playing again, we must begin at beginning of buffer.
+ handle->bufferPointer[0] = 0;
+ }
+
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+ LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ audioPtr = NULL;
+ dataLen = 0;
+
+ stream_.state = STREAM_STOPPED;
+
+ if ( stream_.mode != DUPLEX )
+ MUTEX_LOCK( &stream_.mutex );
+
+ result = buffer->Stop();
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!";
+ errorText_ = errorStream_.str();
+ 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, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+
+ // Zero the DS buffer
+ ZeroMemory( audioPtr, dataLen );
+
+ // Unlock the DS buffer
+ result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+
+ // If we start recording again, we must begin at beginning of buffer.
+ handle->bufferPointer[1] = 0;
+ }
+
+ unlock:
+ timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiDs :: abortStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
+ handle->drainCounter = 2;
+
+ stopStream();
+}
+
+void RtApiDs :: callbackEvent()
+{
+ if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) {
+ Sleep( 50 ); // sleep 50 milliseconds
+ return;
+ }
+
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
+
+ // Check if we were draining the stream and signal is finished.
+ if ( handle->drainCounter > stream_.nBuffers + 2 ) {
+
+ stream_.state = STREAM_STOPPING;
+ if ( handle->internalDrain == false )
+ SetEvent( handle->condition );
+ else
+ stopStream();
+ return;
+ }
+
+ // Invoke user callback to get fresh output data UNLESS we are
+ // draining stream.
+ if ( handle->drainCounter == 0 ) {
+ RtAudioCallback callback = (RtAudioCallback) info->callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
+ handle->xrun[0] = false;
+ }
+ if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
+ status |= RTAUDIO_INPUT_OVERFLOW;
+ handle->xrun[1] = false;
+ }
+ int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, info->userData );
+ if ( cbReturnValue == 2 ) {
+ stream_.state = STREAM_STOPPING;
+ handle->drainCounter = 2;
+ abortStream();
+ return;
+ }
+ else if ( cbReturnValue == 1 ) {
+ handle->drainCounter = 1;
+ handle->internalDrain = true;
+ }
+ }
+
+ HRESULT result;
+ DWORD currentWritePointer, safeWritePointer;
+ DWORD currentReadPointer, safeReadPointer;
+ UINT nextWritePointer;
+
+ LPVOID buffer1 = NULL;
+ LPVOID buffer2 = NULL;
+ DWORD bufferSize1 = 0;
+ DWORD bufferSize2 = 0;
+
+ char *buffer;
+ long bufferBytes;
+
+ MUTEX_LOCK( &stream_.mutex );
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return;
+ }
+
+ if ( buffersRolling == false ) {
+ if ( stream_.mode == DUPLEX ) {
+ //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
+
+ // It takes a while for the devices to get rolling. As a result,
+ // there's no guarantee that the capture and write device pointers
+ // will move in lockstep. Wait here for both devices to start
+ // rolling, and then set our buffer pointers accordingly.
+ // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600
+ // bytes later than the write buffer.
+
+ // Stub: a serious risk of having a pre-emptive scheduling round
+ // take place between the two GetCurrentPosition calls... but I'm
+ // really not sure how to solve the problem. Temporarily boost to
+ // Realtime priority, maybe; but I'm not sure what priority the
+ // DirectSound service threads run at. We *should* be roughly
+ // within a ms or so of correct.
+
+ LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+
+ DWORD startSafeWritePointer, startSafeReadPointer;
+
+ result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ while ( true ) {
+ result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break;
+ Sleep( 1 );
+ }
+
+ //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
+
+ handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
+ if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
+ handle->bufferPointer[1] = safeReadPointer;
+ }
+ else if ( stream_.mode == OUTPUT ) {
+
+ // Set the proper nextWritePosition after initial startup.
+ LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ result = dsWriteBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
+ if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
+ }
+
+ buffersRolling = true;
+ }
+
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+ LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+
+ if ( handle->drainCounter > 1 ) { // write zeros to the output stream
+ bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
+ bufferBytes *= formatBytes( stream_.userFormat );
+ memset( stream_.userBuffer[0], 0, bufferBytes );
+ }
+
+ // Setup parameters and do buffer conversion if necessary.
+ if ( stream_.doConvertBuffer[0] ) {
+ buffer = stream_.deviceBuffer;
+ convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+ bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0];
+ bufferBytes *= formatBytes( stream_.deviceFormat[0] );
+ }
+ else {
+ buffer = stream_.userBuffer[0];
+ bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
+ bufferBytes *= formatBytes( stream_.userFormat );
+ }
+
+ // No byte swapping necessary in DirectSound implementation.
+
+ // Ahhh ... windoze. 16-bit data is signed but 8-bit data is
+ // unsigned. So, we need to convert our signed 8-bit data here to
+ // unsigned.
+ if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 )
+ for ( int i=0; i<bufferBytes; i++ ) buffer[i] = (unsigned char) ( buffer[i] + 128 );
+
+ DWORD dsBufferSize = handle->dsBufferSize[0];
+ nextWritePointer = handle->bufferPointer[0];
+
+ DWORD endWrite, leadPointer;
+ while ( true ) {
+ // Find out where the read and "safe write" pointers are.
+ result = dsBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+
+ // We will copy our output buffer into the region between
+ // safeWritePointer and leadPointer. If leadPointer is not
+ // beyond the next endWrite position, wait until it is.
+ leadPointer = safeWritePointer + handle->dsPointerLeadTime[0];
+ //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl;
+ if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize;
+ if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset
+ endWrite = nextWritePointer + bufferBytes;
+
+ // Check whether the entire write region is behind the play pointer.
+ if ( leadPointer >= endWrite ) break;
+
+ // If we are here, then we must wait until the leadPointer advances
+ // beyond the end of our next write region. We use the
+ // Sleep() function to suspend operation until that happens.
+ double millis = ( endWrite - leadPointer ) * 1000.0;
+ millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);
+ if ( millis < 1.0 ) millis = 1.0;
+ Sleep( (DWORD) millis );
+ }
+
+ if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize )
+ || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) {
+ // We've strayed into the forbidden zone ... resync the read pointer.
+ handle->xrun[0] = true;
+ nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes;
+ if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize;
+ handle->bufferPointer[0] = nextWritePointer;
+ endWrite = nextWritePointer + bufferBytes;
+ }
+
+ // Lock free space in the buffer
+ result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1,
+ &bufferSize1, &buffer2, &bufferSize2, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+
+ // Copy our buffer into the DS buffer
+ CopyMemory( buffer1, buffer, bufferSize1 );
+ if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );
+
+ // Update our buffer offset and unlock sound buffer
+ dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
+ handle->bufferPointer[0] = nextWritePointer;
+ }
+
+ // Don't bother draining input
+ if ( handle->drainCounter ) {
+ handle->drainCounter++;
+ goto unlock;
+ }
+
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+
+ // Setup parameters.
+ if ( stream_.doConvertBuffer[1] ) {
+ buffer = stream_.deviceBuffer;
+ bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1];
+ bufferBytes *= formatBytes( stream_.deviceFormat[1] );
+ }
+ else {
+ buffer = stream_.userBuffer[1];
+ bufferBytes = stream_.bufferSize * stream_.nUserChannels[1];
+ bufferBytes *= formatBytes( stream_.userFormat );
+ }
+
+ LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ long nextReadPointer = handle->bufferPointer[1];
+ DWORD dsBufferSize = handle->dsBufferSize[1];
+
+ // Find out where the write and "safe read" pointers are.
+ result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+
+ if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
+ DWORD endRead = nextReadPointer + bufferBytes;
+
+ // Handling depends on whether we are INPUT or DUPLEX.
+ // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,
+ // then a wait here will drag the write pointers into the forbidden zone.
+ //
+ // In DUPLEX mode, rather than wait, we will back off the read pointer until
+ // it's in a safe position. This causes dropouts, but it seems to be the only
+ // practical way to sync up the read and write pointers reliably, given the
+ // the very complex relationship between phase and increment of the read and write
+ // pointers.
+ //
+ // In order to minimize audible dropouts in DUPLEX mode, we will
+ // provide a pre-roll period of 0.5 seconds in which we return
+ // zeros from the read buffer while the pointers sync up.
+
+ if ( stream_.mode == DUPLEX ) {
+ if ( safeReadPointer < endRead ) {
+ if ( duplexPrerollBytes <= 0 ) {
+ // Pre-roll time over. Be more agressive.
+ int adjustment = endRead-safeReadPointer;
+
+ handle->xrun[1] = true;
+ // Two cases:
+ // - large adjustments: we've probably run out of CPU cycles, so just resync exactly,
+ // and perform fine adjustments later.
+ // - small adjustments: back off by twice as much.
+ if ( adjustment >= 2*bufferBytes )
+ nextReadPointer = safeReadPointer-2*bufferBytes;
+ else
+ nextReadPointer = safeReadPointer-bufferBytes-adjustment;
+
+ if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
+
+ }
+ else {
+ // In pre=roll time. Just do it.
+ nextReadPointer = safeReadPointer - bufferBytes;
+ while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
+ }
+ endRead = nextReadPointer + bufferBytes;
+ }
+ }
+ else { // mode == INPUT
+ while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) {
+ // See comments for playback.
+ double millis = (endRead - safeReadPointer) * 1000.0;
+ millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);
+ if ( millis < 1.0 ) millis = 1.0;
+ Sleep( (DWORD) millis );
+
+ // Wake up and find out where we are now.
+ result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+
+ if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
+ }
+ }
+
+ // Lock free space in the buffer
+ result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1,
+ &bufferSize1, &buffer2, &bufferSize2, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+
+ if ( duplexPrerollBytes <= 0 ) {
+ // Copy our buffer into the DS buffer
+ CopyMemory( buffer, buffer1, bufferSize1 );
+ if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );
+ }
+ else {
+ memset( buffer, 0, bufferSize1 );
+ if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );
+ duplexPrerollBytes -= bufferSize1 + bufferSize2;
+ }
+
+ // Update our buffer offset and unlock sound buffer
+ nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
+ dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ handle->bufferPointer[1] = nextReadPointer;
+
+ // No byte swapping necessary in DirectSound implementation.
+
+ // If necessary, convert 8-bit data from unsigned to signed.
+ if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )
+ for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 );
+
+ // Do buffer conversion if necessary.
+ if ( stream_.doConvertBuffer[1] )
+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+ }
+
+ unlock:
+ MUTEX_UNLOCK( &stream_.mutex );
+ RtApi::tickStreamTime();
+}
+
+// Definitions for utility functions and callbacks
+// specific to the DirectSound implementation.
+
+static unsigned __stdcall callbackHandler( void *ptr )
+{
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiDs *object = (RtApiDs *) info->object;
+ bool* isRunning = &info->isRunning;
+
+ while ( *isRunning == true ) {
+ object->callbackEvent();
+ }
+
+ _endthreadex( 0 );
+ return 0;
+}
+
+static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
+ LPCTSTR description,
+ LPCTSTR /*module*/,
+ LPVOID lpContext )
+{
+ struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext;
+ std::vector<struct DsDevice>& dsDevices = *probeInfo.dsDevices;
+
+ HRESULT hr;
+ bool validDevice = false;
+ if ( probeInfo.isInput == true ) {
+ DSCCAPS caps;
+ LPDIRECTSOUNDCAPTURE object;
+
+ hr = DirectSoundCaptureCreate( lpguid, &object, NULL );
+ 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, NULL );
+ 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 = convertCharPointerToStdString( description );
+ //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" )
+ if ( lpguid == NULL )
+ name = "Default Device";
+ if ( validDevice ) {
+ for ( unsigned int 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( int code )
+{
+ switch ( code ) {
+
+ case DSERR_ALLOCATED:
+ return "Already allocated";
+
+ case DSERR_CONTROLUNAVAIL:
+ return "Control unavailable";
+
+ case DSERR_INVALIDPARAM:
+ return "Invalid parameter";
+
+ case DSERR_INVALIDCALL:
+ return "Invalid call";
+
+ case DSERR_GENERIC:
+ return "Generic error";
+
+ case DSERR_PRIOLEVELNEEDED:
+ return "Priority level needed";
+
+ case DSERR_OUTOFMEMORY:
+ return "Out of memory";
+
+ case DSERR_BADFORMAT:
+ return "The sample rate or the channel format is not supported";
+
+ case DSERR_UNSUPPORTED:
+ return "Not supported";
+
+ case DSERR_NODRIVER:
+ return "No driver";
+
+ case DSERR_ALREADYINITIALIZED:
+ return "Already initialized";
+
+ case DSERR_NOAGGREGATION:
+ return "No aggregation";
+
+ case DSERR_BUFFERLOST:
+ return "Buffer lost";
+
+ case DSERR_OTHERAPPHASPRIO:
+ return "Another application already has priority";
+
+ case DSERR_UNINITIALIZED:
+ return "Uninitialized";
+
+ default:
+ return "DirectSound unknown error";
+ }
+}
+//******************** End of __WINDOWS_DS__ *********************//
+#endif
+
+
+#if defined(__LINUX_ALSA__)
+
+#include <alsa/asoundlib.h>
+#include <unistd.h>
+
+ // A structure to hold various information related to the ALSA API
+ // implementation.
+struct AlsaHandle {
+ snd_pcm_t *handles[2];
+ bool synchronized;
+ bool xrun[2];
+ pthread_cond_t runnable_cv;
+ bool runnable;
+
+ AlsaHandle()
+ :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; }
+};
+
+static void *alsaCallbackHandler( void * ptr );
+
+RtApiAlsa :: RtApiAlsa()
+{
+ // Nothing to do here.
+}
+
+RtApiAlsa :: ~RtApiAlsa()
+{
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
+}
+
+unsigned int RtApiAlsa :: getDeviceCount( void )
+{
+ unsigned nDevices = 0;
+ int result, subdevice, card;
+ char name[64];
+ snd_ctl_t *handle;
+
+ // Count cards and devices
+ card = -1;
+ snd_card_next( &card );
+ while ( card >= 0 ) {
+ sprintf( name, "hw:%d", card );
+ result = snd_ctl_open( &handle, name, 0 );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto nextcard;
+ }
+ subdevice = -1;
+ while( 1 ) {
+ result = snd_ctl_pcm_next_device( handle, &subdevice );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ break;
+ }
+ if ( subdevice < 0 )
+ break;
+ nDevices++;
+ }
+ nextcard:
+ snd_ctl_close( handle );
+ snd_card_next( &card );
+ }
+
+ result = snd_ctl_open( &handle, "default", 0 );
+ if (result == 0) {
+ nDevices++;
+ snd_ctl_close( handle );
+ }
+
+ return nDevices;
+}
+
+RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
+{
+ RtAudio::DeviceInfo info;
+ info.probed = false;
+
+ unsigned nDevices = 0;
+ int result, subdevice, card;
+ char name[64];
+ snd_ctl_t *chandle;
+
+ // Count cards and devices
+ card = -1;
+ subdevice = -1;
+ snd_card_next( &card );
+ while ( card >= 0 ) {
+ sprintf( name, "hw:%d", card );
+ result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto nextcard;
+ }
+ subdevice = -1;
+ while( 1 ) {
+ result = snd_ctl_pcm_next_device( chandle, &subdevice );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ break;
+ }
+ if ( subdevice < 0 ) break;
+ if ( nDevices == device ) {
+ sprintf( name, "hw:%d,%d", card, subdevice );
+ goto foundDevice;
+ }
+ nDevices++;
+ }
+ nextcard:
+ snd_ctl_close( chandle );
+ snd_card_next( &card );
+ }
+
+ result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
+ if ( result == 0 ) {
+ if ( nDevices == device ) {
+ strcpy( name, "default" );
+ goto foundDevice;
+ }
+ nDevices++;
+ }
+
+ if ( nDevices == 0 ) {
+ errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
+ error( RtAudioError::INVALID_USE );
+ return info;
+ }
+
+ if ( device >= nDevices ) {
+ errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
+ error( RtAudioError::INVALID_USE );
+ return info;
+ }
+
+ foundDevice:
+
+ // If a stream is already open, we cannot probe the stream devices.
+ // Thus, use the saved results.
+ if ( stream_.state != STREAM_CLOSED &&
+ ( stream_.device[0] == device || stream_.device[1] == device ) ) {
+ snd_ctl_close( chandle );
+ if ( device >= devices_.size() ) {
+ errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
+ error( RtAudioError::WARNING );
+ return info;
+ }
+ return devices_[ device ];
+ }
+
+ int openMode = SND_PCM_ASYNC;
+ snd_pcm_stream_t stream;
+ snd_pcm_info_t *pcminfo;
+ snd_pcm_info_alloca( &pcminfo );
+ snd_pcm_t *phandle;
+ snd_pcm_hw_params_t *params;
+ snd_pcm_hw_params_alloca( ¶ms );
+
+ // First try for playback unless default device (which has subdev -1)
+ stream = SND_PCM_STREAM_PLAYBACK;
+ snd_pcm_info_set_stream( pcminfo, stream );
+ if ( subdevice != -1 ) {
+ snd_pcm_info_set_device( pcminfo, subdevice );
+ 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;
+ }
+ }
+
+ result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto captureProbe;
+ }
+
+ // The device is open ... fill the parameter structure.
+ result = snd_pcm_hw_params_any( phandle, params );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto captureProbe;
+ }
+
+ // Get output channel information.
+ unsigned int value;
+ result = snd_pcm_hw_params_get_channels_max( params, &value );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto captureProbe;
+ }
+ 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 ) {
+ result = snd_ctl_pcm_info( chandle, pcminfo );
+ snd_ctl_close( chandle );
+ if ( result < 0 ) {
+ // Device probably doesn't support capture.
+ if ( info.outputChannels == 0 ) return info;
+ goto probeParameters;
+ }
+ }
+ else
+ snd_ctl_close( chandle );
+
+ result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ if ( info.outputChannels == 0 ) return info;
+ goto probeParameters;
+ }
+
+ // The device is open ... fill the parameter structure.
+ result = snd_pcm_hw_params_any( phandle, params );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ if ( info.outputChannels == 0 ) return info;
+ goto probeParameters;
+ }
+
+ result = snd_pcm_hw_params_get_channels_max( params, &value );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ if ( info.outputChannels == 0 ) return info;
+ goto probeParameters;
+ }
+ info.inputChannels = value;
+ snd_pcm_close( phandle );
+
+ // 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;
+
+ // ALSA 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;
+
+ 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, name, stream, openMode | SND_PCM_NONBLOCK);
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // The device is open ... fill the parameter structure.
+ result = snd_pcm_hw_params_any( phandle, params );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // Test our discrete set of sample rate values.
+ info.sampleRates.clear();
+ for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
+ if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 ) {
+ info.sampleRates.push_back( SAMPLE_RATES[i] );
+
+ if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
+ info.preferredSampleRate = SAMPLE_RATES[i];
+ }
+ }
+ if ( info.sampleRates.size() == 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // Probe the supported data formats ... we don't care about endian-ness just yet
+ snd_pcm_format_t format;
+ info.nativeFormats = 0;
+ format = SND_PCM_FORMAT_S8;
+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+ info.nativeFormats |= RTAUDIO_SINT8;
+ format = SND_PCM_FORMAT_S16;
+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+ info.nativeFormats |= RTAUDIO_SINT16;
+ format = SND_PCM_FORMAT_S24;
+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+ info.nativeFormats |= RTAUDIO_SINT24;
+ format = SND_PCM_FORMAT_S32;
+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+ info.nativeFormats |= RTAUDIO_SINT32;
+ format = SND_PCM_FORMAT_FLOAT;
+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+ info.nativeFormats |= RTAUDIO_FLOAT32;
+ format = SND_PCM_FORMAT_FLOAT64;
+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+ info.nativeFormats |= RTAUDIO_FLOAT64;
+
+ // Check that we have at least one supported format
+ if ( info.nativeFormats == 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // Get the device name
+ char *cardname;
+ result = snd_card_get_name( card, &cardname );
+ if ( result >= 0 ) {
+ sprintf( name, "hw:%s,%d", cardname, subdevice );
+ free( cardname );
+ }
+ info.name = name;
+
+ // That's all ... close the device and return
+ snd_pcm_close( phandle );
+ info.probed = true;
+ return info;
+}
+
+void RtApiAlsa :: saveDeviceInfo( void )
+{
+ devices_.clear();
+
+ unsigned int nDevices = getDeviceCount();
+ devices_.resize( nDevices );
+ for ( unsigned int i=0; i<nDevices; i++ )
+ devices_[i] = getDeviceInfo( i );
+}
+
+bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options )
+
+{
+#if defined(__RTAUDIO_DEBUG__)
+ snd_output_t *out;
+ snd_output_stdio_attach(&out, stderr, 0);
+#endif
+
+ // I'm not using the "plug" interface ... too much inconsistent behavior.
+
+ unsigned nDevices = 0;
+ int result, subdevice, card;
+ char name[64];
+ snd_ctl_t *chandle;
+
+ if ( options && options->flags & RTAUDIO_ALSA_USE_DEFAULT )
+ snprintf(name, sizeof(name), "%s", "default");
+ else {
+ // Count cards and devices
+ card = -1;
+ snd_card_next( &card );
+ while ( card >= 0 ) {
+ sprintf( name, "hw:%d", card );
+ result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ subdevice = -1;
+ while( 1 ) {
+ result = snd_ctl_pcm_next_device( chandle, &subdevice );
+ if ( result < 0 ) break;
+ if ( subdevice < 0 ) break;
+ if ( nDevices == device ) {
+ sprintf( name, "hw:%d,%d", card, subdevice );
+ snd_ctl_close( chandle );
+ goto foundDevice;
+ }
+ nDevices++;
+ }
+ snd_ctl_close( chandle );
+ snd_card_next( &card );
+ }
+
+ result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
+ if ( result == 0 ) {
+ if ( nDevices == device ) {
+ strcpy( name, "default" );
+ goto foundDevice;
+ }
+ nDevices++;
+ }
+
+ if ( nDevices == 0 ) {
+ // This should not happen because a check is made before this function is called.
+ errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";
+ return FAILURE;
+ }
+
+ if ( device >= nDevices ) {
+ // This should not happen because a check is made before this function is called.
+ errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";
+ return FAILURE;
+ }
+ }
+
+ foundDevice:
+
+ // The getDeviceInfo() function will not work for a device that is
+ // already open. Thus, we'll probe the system before opening a
+ // stream and save the results for use by getDeviceInfo().
+ if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once
+ this->saveDeviceInfo();
+
+ snd_pcm_stream_t stream;
+ if ( mode == OUTPUT )
+ stream = SND_PCM_STREAM_PLAYBACK;
+ else
+ stream = SND_PCM_STREAM_CAPTURE;
+
+ snd_pcm_t *phandle;
+ int openMode = SND_PCM_ASYNC;
+ result = snd_pcm_open( &phandle, name, stream, openMode );
+ if ( result < 0 ) {
+ if ( mode == OUTPUT )
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";
+ else
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Fill the parameter structure.
+ snd_pcm_hw_params_t *hw_params;
+ snd_pcm_hw_params_alloca( &hw_params );
+ result = snd_pcm_hw_params_any( phandle, hw_params );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+#if defined(__RTAUDIO_DEBUG__)
+ fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );
+ snd_pcm_hw_params_dump( hw_params, out );
+#endif
+
+ // Set access ... check user preference.
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {
+ stream_.userInterleaved = false;
+ result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
+ if ( result < 0 ) {
+ result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
+ stream_.deviceInterleaved[mode] = true;
+ }
+ else
+ stream_.deviceInterleaved[mode] = false;
+ }
+ else {
+ stream_.userInterleaved = true;
+ result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
+ if ( result < 0 ) {
+ result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
+ stream_.deviceInterleaved[mode] = false;
+ }
+ else
+ stream_.deviceInterleaved[mode] = true;
+ }
+
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Determine how to set the device format.
+ stream_.userFormat = format;
+ snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;
+
+ if ( format == RTAUDIO_SINT8 )
+ deviceFormat = SND_PCM_FORMAT_S8;
+ else if ( format == RTAUDIO_SINT16 )
+ deviceFormat = SND_PCM_FORMAT_S16;
+ else if ( format == RTAUDIO_SINT24 )
+ deviceFormat = SND_PCM_FORMAT_S24;
+ else if ( format == RTAUDIO_SINT32 )
+ deviceFormat = SND_PCM_FORMAT_S32;
+ else if ( format == RTAUDIO_FLOAT32 )
+ deviceFormat = SND_PCM_FORMAT_FLOAT;
+ else if ( format == RTAUDIO_FLOAT64 )
+ deviceFormat = SND_PCM_FORMAT_FLOAT64;
+
+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {
+ stream_.deviceFormat[mode] = format;
+ goto setFormat;
+ }
+
+ // The user requested format is not natively supported by the device.
+ deviceFormat = SND_PCM_FORMAT_FLOAT64;
+ if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
+ goto setFormat;
+ }
+
+ deviceFormat = SND_PCM_FORMAT_FLOAT;
+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+ goto setFormat;
+ }
+
+ deviceFormat = SND_PCM_FORMAT_S32;
+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ goto setFormat;
+ }
+
+ deviceFormat = SND_PCM_FORMAT_S24;
+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+ goto setFormat;
+ }
+
+ deviceFormat = SND_PCM_FORMAT_S16;
+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ goto setFormat;
+ }
+
+ deviceFormat = SND_PCM_FORMAT_S8;
+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+ goto setFormat;
+ }
+
+ // If we get here, no supported format was found.
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+
+ setFormat:
+ result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Determine whether byte-swaping is necessary.
+ stream_.doByteSwap[mode] = false;
+ if ( deviceFormat != SND_PCM_FORMAT_S8 ) {
+ result = snd_pcm_format_cpu_endian( deviceFormat );
+ if ( result == 0 )
+ stream_.doByteSwap[mode] = true;
+ else if (result < 0) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+
+ // Set the sample rate.
+ result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Determine the number of channels for this device. We support a possible
+ // minimum device channel number > than the value requested by the user.
+ stream_.nUserChannels[mode] = channels;
+ unsigned int value;
+ result = snd_pcm_hw_params_get_channels_max( hw_params, &value );
+ unsigned int deviceChannels = value;
+ if ( result < 0 || deviceChannels < channels + firstChannel ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ result = snd_pcm_hw_params_get_channels_min( hw_params, &value );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ deviceChannels = value;
+ if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;
+ stream_.nDeviceChannels[mode] = deviceChannels;
+
+ // Set the device channels.
+ result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Set the buffer (or period) size.
+ int dir = 0;
+ snd_pcm_uframes_t periodSize = *bufferSize;
+ result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ *bufferSize = periodSize;
+
+ // Set the buffer number, which in ALSA is referred to as the "period".
+ unsigned int periods = 0;
+ if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
+ if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers;
+ if ( periods < 2 ) periods = 4; // a fairly safe default value
+ result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // If attempting to setup a duplex stream, the bufferSize parameter
+ // MUST be the same in both directions!
+ if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ stream_.bufferSize = *bufferSize;
+
+ // Install the hardware configuration
+ result = snd_pcm_hw_params( phandle, hw_params );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+#if defined(__RTAUDIO_DEBUG__)
+ fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
+ snd_pcm_hw_params_dump( hw_params, out );
+#endif
+
+ // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
+ snd_pcm_sw_params_t *sw_params = NULL;
+ snd_pcm_sw_params_alloca( &sw_params );
+ snd_pcm_sw_params_current( phandle, sw_params );
+ snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );
+ snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );
+ snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );
+
+ // The following two settings were suggested by Theo Veenker
+ //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );
+ //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );
+
+ // here are two options for a fix
+ //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );
+ snd_pcm_uframes_t val;
+ snd_pcm_sw_params_get_boundary( sw_params, &val );
+ snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );
+
+ result = snd_pcm_sw_params( phandle, sw_params );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+#if defined(__RTAUDIO_DEBUG__)
+ fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");
+ snd_pcm_sw_params_dump( sw_params, out );
+#endif
+
+ // Set flags for buffer conversion
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
+
+ // Allocate the ApiHandle if necessary and then save.
+ AlsaHandle *apiInfo = 0;
+ if ( stream_.apiHandle == 0 ) {
+ try {
+ apiInfo = (AlsaHandle *) new AlsaHandle;
+ }
+ catch ( std::bad_alloc& ) {
+ errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";
+ goto error;
+ }
+
+ if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) {
+ errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";
+ goto error;
+ }
+
+ stream_.apiHandle = (void *) apiInfo;
+ apiInfo->handles[0] = 0;
+ apiInfo->handles[1] = 0;
+ }
+ else {
+ apiInfo = (AlsaHandle *) stream_.apiHandle;
+ }
+ apiInfo->handles[mode] = phandle;
+ phandle = 0;
+
+ // Allocate necessary internal buffers.
+ unsigned long bufferBytes;
+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
+ }
+
+ if ( stream_.doConvertBuffer[mode] ) {
+
+ bool makeBuffer = true;
+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+ if ( mode == INPUT ) {
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ if ( bufferBytes <= bytesOut ) makeBuffer = false;
+ }
+ }
+
+ if ( makeBuffer ) {
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";
+ goto error;
+ }
+ }
+ }
+
+ stream_.sampleRate = sampleRate;
+ stream_.nBuffers = periods;
+ stream_.device[mode] = device;
+ stream_.state = STREAM_STOPPED;
+
+ // Setup the buffer conversion information structure.
+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+
+ // Setup thread if necessary.
+ if ( stream_.mode == OUTPUT && mode == INPUT ) {
+ // We had already set up an output stream.
+ stream_.mode = DUPLEX;
+ // Link the streams if possible.
+ apiInfo->synchronized = false;
+ if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )
+ apiInfo->synchronized = true;
+ else {
+ errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";
+ error( RtAudioError::WARNING );
+ }
+ }
+ else {
+ stream_.mode = mode;
+
+ // Setup callback thread.
+ stream_.callbackInfo.object = (void *) this;
+
+ // Set the thread attributes for joinable and realtime scheduling
+ // priority (optional). The higher priority will only take affect
+ // if the program is run as root or suid. Note, under Linux
+ // processes with CAP_SYS_NICE privilege, a user can change
+ // scheduling policy and priority (thus need not be root). See
+ // POSIX "capabilities".
+ pthread_attr_t attr;
+ pthread_attr_init( &attr );
+ pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
+
+#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
+ if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
+ // We previously attempted to increase the audio callback priority
+ // to SCHED_RR here via the attributes. However, while no errors
+ // were reported in doing so, it did not work. So, now this is
+ // done in the alsaCallbackHandler function.
+ stream_.callbackInfo.doRealtime = true;
+ int priority = options->priority;
+ int min = sched_get_priority_min( SCHED_RR );
+ int max = sched_get_priority_max( SCHED_RR );
+ if ( priority < min ) priority = min;
+ else if ( priority > max ) priority = max;
+ stream_.callbackInfo.priority = priority;
+ }
+#endif
+
+ stream_.callbackInfo.isRunning = true;
+ result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );
+ pthread_attr_destroy( &attr );
+ if ( result ) {
+ stream_.callbackInfo.isRunning = false;
+ errorText_ = "RtApiAlsa::error creating callback thread!";
+ goto error;
+ }
+ }
+
+ return SUCCESS;
+
+ error:
+ if ( apiInfo ) {
+ pthread_cond_destroy( &apiInfo->runnable_cv );
+ if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
+ if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
+ delete apiInfo;
+ stream_.apiHandle = 0;
+ }
+
+ if ( phandle) snd_pcm_close( phandle );
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ stream_.state = STREAM_CLOSED;
+ return FAILURE;
+}
+
+void RtApiAlsa :: closeStream()
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+ stream_.callbackInfo.isRunning = false;
+ MUTEX_LOCK( &stream_.mutex );
+ if ( stream_.state == STREAM_STOPPED ) {
+ apiInfo->runnable = true;
+ pthread_cond_signal( &apiInfo->runnable_cv );
+ }
+ MUTEX_UNLOCK( &stream_.mutex );
+ pthread_join( stream_.callbackInfo.thread, NULL );
+
+ if ( stream_.state == STREAM_RUNNING ) {
+ stream_.state = STREAM_STOPPED;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
+ snd_pcm_drop( apiInfo->handles[0] );
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
+ snd_pcm_drop( apiInfo->handles[1] );
+ }
+
+ if ( apiInfo ) {
+ pthread_cond_destroy( &apiInfo->runnable_cv );
+ if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
+ if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
+ delete apiInfo;
+ stream_.apiHandle = 0;
+ }
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
+}
+
+void RtApiAlsa :: startStream()
+{
+ // This method calls snd_pcm_prepare if the device isn't already in that state.
+
+ verifyStream();
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ MUTEX_LOCK( &stream_.mutex );
+
+ int result = 0;
+ snd_pcm_state_t state;
+ AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+ snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ state = snd_pcm_state( handle[0] );
+ if ( state != SND_PCM_STATE_PREPARED ) {
+ result = snd_pcm_prepare( handle[0] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+ }
+
+ if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
+ result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open
+ state = snd_pcm_state( handle[1] );
+ if ( state != SND_PCM_STATE_PREPARED ) {
+ result = snd_pcm_prepare( handle[1] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+ }
+
+ stream_.state = STREAM_RUNNING;
+
+ unlock:
+ apiInfo->runnable = true;
+ pthread_cond_signal( &apiInfo->runnable_cv );
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ if ( result >= 0 ) return;
+ error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiAlsa :: stopStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ stream_.state = STREAM_STOPPED;
+ MUTEX_LOCK( &stream_.mutex );
+
+ int result = 0;
+ AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+ snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ if ( apiInfo->synchronized )
+ result = snd_pcm_drop( handle[0] );
+ else
+ result = snd_pcm_drain( handle[0] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
+ result = snd_pcm_drop( handle[1] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ unlock:
+ apiInfo->runnable = false; // fixes high CPU usage when stopped
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ if ( result >= 0 ) return;
+ error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiAlsa :: abortStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ stream_.state = STREAM_STOPPED;
+ MUTEX_LOCK( &stream_.mutex );
+
+ int result = 0;
+ AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+ snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ result = snd_pcm_drop( handle[0] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
+ result = snd_pcm_drop( handle[1] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ unlock:
+ apiInfo->runnable = false; // fixes high CPU usage when stopped
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ if ( result >= 0 ) return;
+ error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiAlsa :: callbackEvent()
+{
+ AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_LOCK( &stream_.mutex );
+ while ( !apiInfo->runnable )
+ pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex );
+
+ if ( stream_.state != STREAM_RUNNING ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return;
+ }
+ MUTEX_UNLOCK( &stream_.mutex );
+ }
+
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ int doStopStream = 0;
+ RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {
+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
+ apiInfo->xrun[0] = false;
+ }
+ if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {
+ status |= RTAUDIO_INPUT_OVERFLOW;
+ apiInfo->xrun[1] = false;
+ }
+ doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
+
+ if ( doStopStream == 2 ) {
+ abortStream();
+ return;
+ }
+
+ MUTEX_LOCK( &stream_.mutex );
+
+ // The state might change while waiting on a mutex.
+ if ( stream_.state == STREAM_STOPPED ) goto unlock;
+
+ int result;
+ char *buffer;
+ int channels;
+ snd_pcm_t **handle;
+ snd_pcm_sframes_t frames;
+ RtAudioFormat format;
+ handle = (snd_pcm_t **) apiInfo->handles;
+
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+
+ // Setup parameters.
+ if ( stream_.doConvertBuffer[1] ) {
+ buffer = stream_.deviceBuffer;
+ channels = stream_.nDeviceChannels[1];
+ format = stream_.deviceFormat[1];
+ }
+ else {
+ buffer = stream_.userBuffer[1];
+ channels = stream_.nUserChannels[1];
+ format = stream_.userFormat;
+ }
+
+ // Read samples from device in interleaved/non-interleaved format.
+ if ( stream_.deviceInterleaved[1] )
+ result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize );
+ else {
+ void *bufs[channels];
+ size_t offset = stream_.bufferSize * formatBytes( format );
+ for ( int i=0; i<channels; i++ )
+ bufs[i] = (void *) (buffer + (i * offset));
+ result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize );
+ }
+
+ if ( result < (int) stream_.bufferSize ) {
+ // Either an error or overrun occured.
+ if ( result == -EPIPE ) {
+ snd_pcm_state_t state = snd_pcm_state( handle[1] );
+ if ( state == SND_PCM_STATE_XRUN ) {
+ apiInfo->xrun[1] = true;
+ result = snd_pcm_prepare( handle[1] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ }
+ }
+ else {
+ errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ }
+ }
+ else {
+ errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ }
+ error( RtAudioError::WARNING );
+ goto tryOutput;
+ }
+
+ // Do byte swapping if necessary.
+ if ( stream_.doByteSwap[1] )
+ byteSwapBuffer( buffer, stream_.bufferSize * channels, format );
+
+ // Do buffer conversion if necessary.
+ if ( stream_.doConvertBuffer[1] )
+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+
+ // Check stream latency
+ result = snd_pcm_delay( handle[1], &frames );
+ if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;
+ }
+
+ tryOutput:
+
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+ // Setup parameters and do buffer conversion if necessary.
+ if ( stream_.doConvertBuffer[0] ) {
+ buffer = stream_.deviceBuffer;
+ convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+ channels = stream_.nDeviceChannels[0];
+ format = stream_.deviceFormat[0];
+ }
+ else {
+ buffer = stream_.userBuffer[0];
+ channels = stream_.nUserChannels[0];
+ format = stream_.userFormat;
+ }
+
+ // Do byte swapping if necessary.
+ if ( stream_.doByteSwap[0] )
+ byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
+
+ // Write samples to device in interleaved/non-interleaved format.
+ if ( stream_.deviceInterleaved[0] )
+ result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );
+ else {
+ void *bufs[channels];
+ size_t offset = stream_.bufferSize * formatBytes( format );
+ for ( int i=0; i<channels; i++ )
+ bufs[i] = (void *) (buffer + (i * offset));
+ result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );
+ }
+
+ if ( result < (int) stream_.bufferSize ) {
+ // Either an error or underrun occured.
+ if ( result == -EPIPE ) {
+ snd_pcm_state_t state = snd_pcm_state( handle[0] );
+ if ( state == SND_PCM_STATE_XRUN ) {
+ apiInfo->xrun[0] = true;
+ result = snd_pcm_prepare( handle[0] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ }
+ else
+ errorText_ = "RtApiAlsa::callbackEvent: audio write error, underrun.";
+ }
+ else {
+ errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ }
+ }
+ else {
+ errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ }
+ error( RtAudioError::WARNING );
+ goto unlock;
+ }
+
+ // Check stream latency
+ result = snd_pcm_delay( handle[0], &frames );
+ if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;
+ }
+
+ unlock:
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ RtApi::tickStreamTime();
+ if ( doStopStream == 1 ) this->stopStream();
+}
+
+static void *alsaCallbackHandler( void *ptr )
+{
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiAlsa *object = (RtApiAlsa *) info->object;
+ bool *isRunning = &info->isRunning;
+
+#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
+ if ( info->doRealtime ) {
+ pthread_t tID = pthread_self(); // ID of this thread
+ sched_param prio = { info->priority }; // scheduling priority of thread
+ pthread_setschedparam( tID, SCHED_RR, &prio );
+ }
+#endif
+
+ while ( *isRunning == true ) {
+ pthread_testcancel();
+ object->callbackEvent();
+ }
+
+ pthread_exit( NULL );
+}
+
+//******************** End of __LINUX_ALSA__ *********************//
+#endif
+
+#if defined(__LINUX_PULSE__)
+
+// Code written by Peter Meerwald, pmeerw at pmeerw.net
+// and Tristan Matthews.
+
+#include <pulse/error.h>
+#include <pulse/simple.h>
+#include <cstdio>
+
+static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000,
+ 44100, 48000, 96000, 0};
+
+struct rtaudio_pa_format_mapping_t {
+ RtAudioFormat rtaudio_format;
+ pa_sample_format_t pa_format;
+};
+
+static const rtaudio_pa_format_mapping_t supported_sampleformats[] = {
+ {RTAUDIO_SINT16, PA_SAMPLE_S16LE},
+ {RTAUDIO_SINT32, PA_SAMPLE_S32LE},
+ {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE},
+ {0, PA_SAMPLE_INVALID}};
+
+struct PulseAudioHandle {
+ pa_simple *s_play;
+ pa_simple *s_rec;
+ pthread_t thread;
+ pthread_cond_t runnable_cv;
+ bool runnable;
+ PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { }
+};
+
+RtApiPulse::~RtApiPulse()
+{
+ if ( stream_.state != STREAM_CLOSED )
+ closeStream();
+}
+
+unsigned int RtApiPulse::getDeviceCount( void )
+{
+ return 1;
+}
+
+RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ )
+{
+ RtAudio::DeviceInfo info;
+ info.probed = true;
+ info.name = "PulseAudio";
+ info.outputChannels = 2;
+ info.inputChannels = 2;
+ info.duplexChannels = 2;
+ info.isDefaultOutput = true;
+ info.isDefaultInput = true;
+
+ for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
+ info.sampleRates.push_back( *sr );
+
+ info.preferredSampleRate = 48000;
+ info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32;
+
+ return info;
+}
+
+static void *pulseaudio_callback( void * user )
+{
+ CallbackInfo *cbi = static_cast<CallbackInfo *>( user );
+ RtApiPulse *context = static_cast<RtApiPulse *>( cbi->object );
+ volatile bool *isRunning = &cbi->isRunning;
+
+ while ( *isRunning ) {
+ pthread_testcancel();
+ context->callbackEvent();
+ }
+
+ pthread_exit( NULL );
+}
+
+void RtApiPulse::closeStream( void )
+{
+ PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
+
+ stream_.callbackInfo.isRunning = false;
+ if ( pah ) {
+ MUTEX_LOCK( &stream_.mutex );
+ if ( stream_.state == STREAM_STOPPED ) {
+ pah->runnable = true;
+ pthread_cond_signal( &pah->runnable_cv );
+ }
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ pthread_join( pah->thread, 0 );
+ if ( pah->s_play ) {
+ pa_simple_flush( pah->s_play, NULL );
+ pa_simple_free( pah->s_play );
+ }
+ if ( pah->s_rec )
+ pa_simple_free( pah->s_rec );
+
+ pthread_cond_destroy( &pah->runnable_cv );
+ delete pah;
+ stream_.apiHandle = 0;
+ }
+
+ if ( stream_.userBuffer[0] ) {
+ free( stream_.userBuffer[0] );
+ stream_.userBuffer[0] = 0;
+ }
+ if ( stream_.userBuffer[1] ) {
+ free( stream_.userBuffer[1] );
+ stream_.userBuffer[1] = 0;
+ }
+
+ stream_.state = STREAM_CLOSED;
+ stream_.mode = UNINITIALIZED;
+}
+
+void RtApiPulse::callbackEvent( void )
+{
+ PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
+
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_LOCK( &stream_.mutex );
+ while ( !pah->runnable )
+ pthread_cond_wait( &pah->runnable_cv, &stream_.mutex );
+
+ if ( stream_.state != STREAM_RUNNING ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return;
+ }
+ MUTEX_UNLOCK( &stream_.mutex );
+ }
+
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... "
+ "this shouldn't happen!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT],
+ stream_.bufferSize, streamTime, status,
+ stream_.callbackInfo.userData );
+
+ if ( doStopStream == 2 ) {
+ abortStream();
+ return;
+ }
+
+ MUTEX_LOCK( &stream_.mutex );
+ void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT];
+ void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT];
+
+ if ( stream_.state != STREAM_RUNNING )
+ goto unlock;
+
+ int pa_error;
+ size_t bytes;
+ if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ if ( stream_.doConvertBuffer[OUTPUT] ) {
+ convertBuffer( stream_.deviceBuffer,
+ stream_.userBuffer[OUTPUT],
+ stream_.convertInfo[OUTPUT] );
+ bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize *
+ formatBytes( stream_.deviceFormat[OUTPUT] );
+ } else
+ bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize *
+ formatBytes( stream_.userFormat );
+
+ if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) {
+ errorStream_ << "RtApiPulse::callbackEvent: audio write error, " <<
+ pa_strerror( pa_error ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ }
+ }
+
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX) {
+ if ( stream_.doConvertBuffer[INPUT] )
+ bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize *
+ formatBytes( stream_.deviceFormat[INPUT] );
+ else
+ bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize *
+ formatBytes( stream_.userFormat );
+
+ if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) {
+ errorStream_ << "RtApiPulse::callbackEvent: audio read error, " <<
+ pa_strerror( pa_error ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ }
+ if ( stream_.doConvertBuffer[INPUT] ) {
+ convertBuffer( stream_.userBuffer[INPUT],
+ stream_.deviceBuffer,
+ stream_.convertInfo[INPUT] );
+ }
+ }
+
+ unlock:
+ MUTEX_UNLOCK( &stream_.mutex );
+ RtApi::tickStreamTime();
+
+ if ( doStopStream == 1 )
+ stopStream();
+}
+
+void RtApiPulse::startStream( void )
+{
+ PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
+
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiPulse::startStream(): the stream is not open!";
+ error( RtAudioError::INVALID_USE );
+ return;
+ }
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiPulse::startStream(): the stream is already running!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ MUTEX_LOCK( &stream_.mutex );
+
+ stream_.state = STREAM_RUNNING;
+
+ pah->runnable = true;
+ pthread_cond_signal( &pah->runnable_cv );
+ MUTEX_UNLOCK( &stream_.mutex );
+}
+
+void RtApiPulse::stopStream( void )
+{
+ PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
+
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiPulse::stopStream(): the stream is not open!";
+ error( RtAudioError::INVALID_USE );
+ return;
+ }
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ stream_.state = STREAM_STOPPED;
+ MUTEX_LOCK( &stream_.mutex );
+
+ if ( pah && pah->s_play ) {
+ int pa_error;
+ if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) {
+ errorStream_ << "RtApiPulse::stopStream: error draining output device, " <<
+ pa_strerror( pa_error ) << ".";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ }
+
+ stream_.state = STREAM_STOPPED;
+ MUTEX_UNLOCK( &stream_.mutex );
+}
+
+void RtApiPulse::abortStream( void )
+{
+ PulseAudioHandle *pah = static_cast<PulseAudioHandle*>( stream_.apiHandle );
+
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiPulse::abortStream(): the stream is not open!";
+ error( RtAudioError::INVALID_USE );
+ return;
+ }
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ stream_.state = STREAM_STOPPED;
+ MUTEX_LOCK( &stream_.mutex );
+
+ if ( pah && pah->s_play ) {
+ int pa_error;
+ if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) {
+ errorStream_ << "RtApiPulse::abortStream: error flushing output device, " <<
+ pa_strerror( pa_error ) << ".";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ }
+
+ stream_.state = STREAM_STOPPED;
+ MUTEX_UNLOCK( &stream_.mutex );
+}
+
+bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
+ unsigned int channels, unsigned int firstChannel,
+ unsigned int sampleRate, RtAudioFormat format,
+ unsigned int *bufferSize, RtAudio::StreamOptions *options )
+{
+ PulseAudioHandle *pah = 0;
+ unsigned long bufferBytes = 0;
+ pa_sample_spec ss;
+
+ if ( device != 0 ) return false;
+ if ( mode != INPUT && mode != OUTPUT ) return false;
+ if ( channels != 1 && channels != 2 ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels.";
+ return false;
+ }
+ ss.channels = channels;
+
+ if ( firstChannel != 0 ) return false;
+
+ bool sr_found = false;
+ for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) {
+ if ( sampleRate == *sr ) {
+ sr_found = true;
+ stream_.sampleRate = sampleRate;
+ ss.rate = sampleRate;
+ break;
+ }
+ }
+ if ( !sr_found ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate.";
+ return false;
+ }
+
+ bool sf_found = 0;
+ for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats;
+ sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) {
+ if ( format == sf->rtaudio_format ) {
+ sf_found = true;
+ stream_.userFormat = sf->rtaudio_format;
+ stream_.deviceFormat[mode] = stream_.userFormat;
+ ss.format = sf->pa_format;
+ break;
+ }
+ }
+ if ( !sf_found ) { // Use internal data format conversion.
+ stream_.userFormat = format;
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+ ss.format = PA_SAMPLE_FLOAT32LE;
+ }
+
+ // Set other stream parameters.
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+ else stream_.userInterleaved = true;
+ stream_.deviceInterleaved[mode] = true;
+ stream_.nBuffers = 1;
+ stream_.doByteSwap[mode] = false;
+ stream_.nUserChannels[mode] = channels;
+ stream_.nDeviceChannels[mode] = channels + firstChannel;
+ stream_.channelOffset[mode] = 0;
+ std::string streamName = "RtAudio";
+
+ // Set flags for buffer conversion.
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
+ stream_.doConvertBuffer[mode] = true;
+
+ // Allocate necessary internal buffers.
+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
+ }
+ stream_.bufferSize = *bufferSize;
+
+ if ( stream_.doConvertBuffer[mode] ) {
+
+ bool makeBuffer = true;
+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+ if ( mode == INPUT ) {
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ if ( bufferBytes <= bytesOut ) makeBuffer = false;
+ }
+ }
+
+ if ( makeBuffer ) {
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory.";
+ goto error;
+ }
+ }
+ }
+
+ stream_.device[mode] = device;
+
+ // Setup the buffer conversion information structure.
+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+
+ if ( !stream_.apiHandle ) {
+ PulseAudioHandle *pah = new PulseAudioHandle;
+ if ( !pah ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle.";
+ goto error;
+ }
+
+ stream_.apiHandle = pah;
+ if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable.";
+ goto error;
+ }
+ }
+ pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
+
+ int error;
+ if ( options && !options->streamName.empty() ) streamName = options->streamName;
+ switch ( mode ) {
+ case INPUT:
+ pa_buffer_attr buffer_attr;
+ buffer_attr.fragsize = bufferBytes;
+ buffer_attr.maxlength = -1;
+
+ pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &error );
+ if ( !pah->s_rec ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server.";
+ goto error;
+ }
+ break;
+ case OUTPUT:
+ pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error );
+ if ( !pah->s_play ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";
+ goto error;
+ }
+ break;
+ default:
+ goto error;
+ }
+
+ if ( stream_.mode == UNINITIALIZED )
+ stream_.mode = mode;
+ else if ( stream_.mode == mode )
+ goto error;
+ else
+ stream_.mode = DUPLEX;
+
+ if ( !stream_.callbackInfo.isRunning ) {
+ stream_.callbackInfo.object = this;
+ stream_.callbackInfo.isRunning = true;
+ if ( pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo) != 0 ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread.";
+ goto error;
+ }
+ }
+
+ stream_.state = STREAM_STOPPED;
+ return true;
+
+ error:
+ if ( pah && stream_.callbackInfo.isRunning ) {
+ pthread_cond_destroy( &pah->runnable_cv );
+ delete pah;
+ stream_.apiHandle = 0;
+ }
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ return FAILURE;
+}
+
+//******************** End of __LINUX_PULSE__ *********************//
+#endif
+
+#if defined(__LINUX_OSS__)
+
+#include <unistd.h>
+#include <sys/ioctl.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <sys/soundcard.h>
+#include <errno.h>
+#include <math.h>
+
+static void *ossCallbackHandler(void * ptr);
+
+// A structure to hold various information related to the OSS API
+// implementation.
+struct OssHandle {
+ int id[2]; // device ids
+ bool xrun[2];
+ bool triggered;
+ pthread_cond_t runnable;
+
+ OssHandle()
+ :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
+};
+
+RtApiOss :: RtApiOss()
+{
+ // Nothing to do here.
+}
+
+RtApiOss :: ~RtApiOss()
+{
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
+}
+
+unsigned int RtApiOss :: getDeviceCount( void )
+{
+ int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
+ if ( mixerfd == -1 ) {
+ errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";
+ error( RtAudioError::WARNING );
+ return 0;
+ }
+
+ oss_sysinfo sysinfo;
+ if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {
+ close( mixerfd );
+ errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";
+ error( RtAudioError::WARNING );
+ return 0;
+ }
+
+ close( mixerfd );
+ return sysinfo.numaudios;
+}
+
+RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
+{
+ RtAudio::DeviceInfo info;
+ info.probed = false;
+
+ int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
+ if ( mixerfd == -1 ) {
+ errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ oss_sysinfo sysinfo;
+ int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
+ if ( result == -1 ) {
+ close( mixerfd );
+ errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ unsigned nDevices = sysinfo.numaudios;
+ if ( nDevices == 0 ) {
+ close( mixerfd );
+ errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
+ error( RtAudioError::INVALID_USE );
+ return info;
+ }
+
+ if ( device >= nDevices ) {
+ close( mixerfd );
+ errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
+ error( RtAudioError::INVALID_USE );
+ return info;
+ }
+
+ oss_audioinfo ainfo;
+ ainfo.dev = device;
+ result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
+ close( mixerfd );
+ if ( result == -1 ) {
+ errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // Probe channels
+ if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;
+ if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;
+ if ( ainfo.caps & PCM_CAP_DUPLEX ) {
+ if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )
+ info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+ }
+
+ // Probe data formats ... do for input
+ unsigned long mask = ainfo.iformats;
+ if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )
+ info.nativeFormats |= RTAUDIO_SINT16;
+ if ( mask & AFMT_S8 )
+ info.nativeFormats |= RTAUDIO_SINT8;
+ if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )
+ info.nativeFormats |= RTAUDIO_SINT32;
+ if ( mask & AFMT_FLOAT )
+ info.nativeFormats |= RTAUDIO_FLOAT32;
+ if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )
+ info.nativeFormats |= RTAUDIO_SINT24;
+
+ // Check that we have at least one supported format
+ if ( info.nativeFormats == 0 ) {
+ errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // Probe the supported sample rates.
+ info.sampleRates.clear();
+ if ( ainfo.nrates ) {
+ for ( unsigned int i=0; i<ainfo.nrates; i++ ) {
+ for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
+ if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {
+ info.sampleRates.push_back( SAMPLE_RATES[k] );
+
+ if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
+ info.preferredSampleRate = SAMPLE_RATES[k];
+
+ break;
+ }
+ }
+ }
+ }
+ else {
+ // Check min and max rate values;
+ for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
+ if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] ) {
+ info.sampleRates.push_back( SAMPLE_RATES[k] );
+
+ if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
+ info.preferredSampleRate = SAMPLE_RATES[k];
+ }
+ }
+ }
+
+ if ( info.sampleRates.size() == 0 ) {
+ errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ }
+ else {
+ info.probed = true;
+ info.name = ainfo.name;
+ }
+
+ return info;
+}
+
+
+bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options )
+{
+ int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
+ if ( mixerfd == -1 ) {
+ errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";
+ return FAILURE;
+ }
+
+ oss_sysinfo sysinfo;
+ int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
+ if ( result == -1 ) {
+ close( mixerfd );
+ errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";
+ return FAILURE;
+ }
+
+ unsigned nDevices = sysinfo.numaudios;
+ if ( nDevices == 0 ) {
+ // This should not happen because a check is made before this function is called.
+ close( mixerfd );
+ errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";
+ return FAILURE;
+ }
+
+ if ( device >= nDevices ) {
+ // This should not happen because a check is made before this function is called.
+ close( mixerfd );
+ errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";
+ return FAILURE;
+ }
+
+ oss_audioinfo ainfo;
+ ainfo.dev = device;
+ result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
+ close( mixerfd );
+ if ( result == -1 ) {
+ errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Check if device supports input or output
+ if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||
+ ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {
+ if ( mode == OUTPUT )
+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";
+ else
+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ int flags = 0;
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ if ( mode == OUTPUT )
+ flags |= O_WRONLY;
+ else { // mode == INPUT
+ if (stream_.mode == OUTPUT && stream_.device[0] == device) {
+ // We just set the same device for playback ... close and reopen for duplex (OSS only).
+ close( handle->id[0] );
+ handle->id[0] = 0;
+ if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {
+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ // Check that the number previously set channels is the same.
+ if ( stream_.nUserChannels[0] != channels ) {
+ errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ flags |= O_RDWR;
+ }
+ else
+ flags |= O_RDONLY;
+ }
+
+ // Set exclusive access if specified.
+ if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;
+
+ // Try to open the device.
+ int fd;
+ fd = open( ainfo.devnode, flags, 0 );
+ if ( fd == -1 ) {
+ if ( errno == EBUSY )
+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";
+ else
+ errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // For duplex operation, specifically set this mode (this doesn't seem to work).
+ /*
+ if ( flags | O_RDWR ) {
+ result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );
+ if ( result == -1) {
+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+ */
+
+ // Check the device channel support.
+ stream_.nUserChannels[mode] = channels;
+ if ( ainfo.max_channels < (int)(channels + firstChannel) ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Set the number of channels.
+ int deviceChannels = channels + firstChannel;
+ result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );
+ if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ stream_.nDeviceChannels[mode] = deviceChannels;
+
+ // Get the data format mask
+ int mask;
+ result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );
+ if ( result == -1 ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Determine how to set the device format.
+ stream_.userFormat = format;
+ int deviceFormat = -1;
+ stream_.doByteSwap[mode] = false;
+ if ( format == RTAUDIO_SINT8 ) {
+ if ( mask & AFMT_S8 ) {
+ deviceFormat = AFMT_S8;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+ }
+ }
+ else if ( format == RTAUDIO_SINT16 ) {
+ if ( mask & AFMT_S16_NE ) {
+ deviceFormat = AFMT_S16_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ }
+ else if ( mask & AFMT_S16_OE ) {
+ deviceFormat = AFMT_S16_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ stream_.doByteSwap[mode] = true;
+ }
+ }
+ else if ( format == RTAUDIO_SINT24 ) {
+ if ( mask & AFMT_S24_NE ) {
+ deviceFormat = AFMT_S24_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+ }
+ else if ( mask & AFMT_S24_OE ) {
+ deviceFormat = AFMT_S24_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+ stream_.doByteSwap[mode] = true;
+ }
+ }
+ else if ( format == RTAUDIO_SINT32 ) {
+ if ( mask & AFMT_S32_NE ) {
+ deviceFormat = AFMT_S32_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ }
+ else if ( mask & AFMT_S32_OE ) {
+ deviceFormat = AFMT_S32_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ stream_.doByteSwap[mode] = true;
+ }
+ }
+
+ if ( deviceFormat == -1 ) {
+ // The user requested format is not natively supported by the device.
+ if ( mask & AFMT_S16_NE ) {
+ deviceFormat = AFMT_S16_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ }
+ else if ( mask & AFMT_S32_NE ) {
+ deviceFormat = AFMT_S32_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ }
+ else if ( mask & AFMT_S24_NE ) {
+ deviceFormat = AFMT_S24_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+ }
+ else if ( mask & AFMT_S16_OE ) {
+ deviceFormat = AFMT_S16_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ stream_.doByteSwap[mode] = true;
+ }
+ else if ( mask & AFMT_S32_OE ) {
+ deviceFormat = AFMT_S32_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ stream_.doByteSwap[mode] = true;
+ }
+ else if ( mask & AFMT_S24_OE ) {
+ deviceFormat = AFMT_S24_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+ stream_.doByteSwap[mode] = true;
+ }
+ else if ( mask & AFMT_S8) {
+ deviceFormat = AFMT_S8;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+ }
+ }
+
+ if ( stream_.deviceFormat[mode] == 0 ) {
+ // This really shouldn't happen ...
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Set the data format.
+ int temp = deviceFormat;
+ result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );
+ if ( result == -1 || deviceFormat != temp ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Attempt to set the buffer size. According to OSS, the minimum
+ // number of buffers is two. The supposed minimum buffer size is 16
+ // bytes, so that will be our lower bound. The argument to this
+ // call is in the form 0xMMMMSSSS (hex), where the buffer size (in
+ // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
+ // We'll check the actual value used near the end of the setup
+ // procedure.
+ int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;
+ if ( ossBufferBytes < 16 ) ossBufferBytes = 16;
+ int buffers = 0;
+ if ( options ) buffers = options->numberOfBuffers;
+ if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;
+ if ( buffers < 2 ) buffers = 3;
+ temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );
+ result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );
+ if ( result == -1 ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ stream_.nBuffers = buffers;
+
+ // Save buffer size (in sample frames).
+ *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );
+ stream_.bufferSize = *bufferSize;
+
+ // Set the sample rate.
+ int srate = sampleRate;
+ result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );
+ if ( result == -1 ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Verify the sample rate setup worked.
+ if ( abs( srate - sampleRate ) > 100 ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ stream_.sampleRate = sampleRate;
+
+ if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) {
+ // We're doing duplex setup here.
+ stream_.deviceFormat[0] = stream_.deviceFormat[1];
+ stream_.nDeviceChannels[0] = deviceChannels;
+ }
+
+ // Set interleaving parameters.
+ stream_.userInterleaved = true;
+ stream_.deviceInterleaved[mode] = true;
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
+ stream_.userInterleaved = false;
+
+ // Set flags for buffer conversion
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
+
+ // Allocate the stream handles if necessary and then save.
+ if ( stream_.apiHandle == 0 ) {
+ try {
+ handle = new OssHandle;
+ }
+ catch ( std::bad_alloc& ) {
+ errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";
+ goto error;
+ }
+
+ if ( pthread_cond_init( &handle->runnable, NULL ) ) {
+ errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";
+ goto error;
+ }
+
+ stream_.apiHandle = (void *) handle;
+ }
+ else {
+ handle = (OssHandle *) stream_.apiHandle;
+ }
+ handle->id[mode] = fd;
+
+ // Allocate necessary internal buffers.
+ unsigned long bufferBytes;
+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
+ }
+
+ if ( stream_.doConvertBuffer[mode] ) {
+
+ bool makeBuffer = true;
+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+ if ( mode == INPUT ) {
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ if ( bufferBytes <= bytesOut ) makeBuffer = false;
+ }
+ }
+
+ if ( makeBuffer ) {
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";
+ goto error;
+ }
+ }
+ }
+
+ stream_.device[mode] = device;
+ stream_.state = STREAM_STOPPED;
+
+ // Setup the buffer conversion information structure.
+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+
+ // Setup thread if necessary.
+ if ( stream_.mode == OUTPUT && mode == INPUT ) {
+ // We had already set up an output stream.
+ stream_.mode = DUPLEX;
+ if ( stream_.device[0] == device ) handle->id[0] = fd;
+ }
+ else {
+ stream_.mode = mode;
+
+ // Setup callback thread.
+ stream_.callbackInfo.object = (void *) this;
+
+ // Set the thread attributes for joinable and realtime scheduling
+ // priority. The higher priority will only take affect if the
+ // program is run as root or suid.
+ pthread_attr_t attr;
+ pthread_attr_init( &attr );
+ pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
+#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
+ if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
+ struct sched_param param;
+ int priority = options->priority;
+ int min = sched_get_priority_min( SCHED_RR );
+ int max = sched_get_priority_max( SCHED_RR );
+ if ( priority < min ) priority = min;
+ else if ( priority > max ) priority = max;
+ param.sched_priority = priority;
+ pthread_attr_setschedparam( &attr, ¶m );
+ pthread_attr_setschedpolicy( &attr, SCHED_RR );
+ }
+ else
+ pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#else
+ pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#endif
+
+ stream_.callbackInfo.isRunning = true;
+ result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );
+ pthread_attr_destroy( &attr );
+ if ( result ) {
+ stream_.callbackInfo.isRunning = false;
+ errorText_ = "RtApiOss::error creating callback thread!";
+ goto error;
+ }
+ }
+
+ return SUCCESS;
+
+ error:
+ if ( handle ) {
+ pthread_cond_destroy( &handle->runnable );
+ if ( handle->id[0] ) close( handle->id[0] );
+ if ( handle->id[1] ) close( handle->id[1] );
+ delete handle;
+ stream_.apiHandle = 0;
+ }
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ return FAILURE;
+}
+
+void RtApiOss :: closeStream()
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiOss::closeStream(): no open stream to close!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ stream_.callbackInfo.isRunning = false;
+ MUTEX_LOCK( &stream_.mutex );
+ if ( stream_.state == STREAM_STOPPED )
+ pthread_cond_signal( &handle->runnable );
+ MUTEX_UNLOCK( &stream_.mutex );
+ pthread_join( stream_.callbackInfo.thread, NULL );
+
+ if ( stream_.state == STREAM_RUNNING ) {
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
+ ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
+ else
+ ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
+ stream_.state = STREAM_STOPPED;
+ }
+
+ if ( handle ) {
+ pthread_cond_destroy( &handle->runnable );
+ if ( handle->id[0] ) close( handle->id[0] );
+ if ( handle->id[1] ) close( handle->id[1] );
+ delete handle;
+ stream_.apiHandle = 0;
+ }
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
+}
+
+void RtApiOss :: startStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiOss::startStream(): the stream is already running!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ MUTEX_LOCK( &stream_.mutex );
+
+ stream_.state = STREAM_RUNNING;
+
+ // No need to do anything else here ... OSS automatically starts
+ // when fed samples.
+
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ pthread_cond_signal( &handle->runnable );
+}
+
+void RtApiOss :: stopStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ MUTEX_LOCK( &stream_.mutex );
+
+ // The state might change while waiting on a mutex.
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return;
+ }
+
+ int result = 0;
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+ // Flush the output with zeros a few times.
+ char *buffer;
+ int samples;
+ RtAudioFormat format;
+
+ if ( stream_.doConvertBuffer[0] ) {
+ buffer = stream_.deviceBuffer;
+ samples = stream_.bufferSize * stream_.nDeviceChannels[0];
+ format = stream_.deviceFormat[0];
+ }
+ else {
+ buffer = stream_.userBuffer[0];
+ samples = stream_.bufferSize * stream_.nUserChannels[0];
+ format = stream_.userFormat;
+ }
+
+ memset( buffer, 0, samples * formatBytes(format) );
+ for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {
+ result = write( handle->id[0], buffer, samples * formatBytes(format) );
+ if ( result == -1 ) {
+ errorText_ = "RtApiOss::stopStream: audio write error.";
+ error( RtAudioError::WARNING );
+ }
+ }
+
+ result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
+ if ( result == -1 ) {
+ errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ handle->triggered = false;
+ }
+
+ if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
+ result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
+ if ( result == -1 ) {
+ errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ unlock:
+ stream_.state = STREAM_STOPPED;
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ if ( result != -1 ) return;
+ error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiOss :: abortStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ MUTEX_LOCK( &stream_.mutex );
+
+ // The state might change while waiting on a mutex.
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return;
+ }
+
+ int result = 0;
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
+ if ( result == -1 ) {
+ errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ handle->triggered = false;
+ }
+
+ if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
+ result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
+ if ( result == -1 ) {
+ errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ unlock:
+ stream_.state = STREAM_STOPPED;
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ if ( result != -1 ) return;
+ error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiOss :: callbackEvent()
+{
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_LOCK( &stream_.mutex );
+ pthread_cond_wait( &handle->runnable, &stream_.mutex );
+ if ( stream_.state != STREAM_RUNNING ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return;
+ }
+ MUTEX_UNLOCK( &stream_.mutex );
+ }
+
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ // Invoke user callback to get fresh output data.
+ int doStopStream = 0;
+ RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
+ handle->xrun[0] = false;
+ }
+ if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
+ status |= RTAUDIO_INPUT_OVERFLOW;
+ handle->xrun[1] = false;
+ }
+ doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
+ if ( doStopStream == 2 ) {
+ this->abortStream();
+ return;
+ }
+
+ MUTEX_LOCK( &stream_.mutex );
+
+ // The state might change while waiting on a mutex.
+ if ( stream_.state == STREAM_STOPPED ) goto unlock;
+
+ int result;
+ char *buffer;
+ int samples;
+ RtAudioFormat format;
+
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+ // Setup parameters and do buffer conversion if necessary.
+ if ( stream_.doConvertBuffer[0] ) {
+ buffer = stream_.deviceBuffer;
+ convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+ samples = stream_.bufferSize * stream_.nDeviceChannels[0];
+ format = stream_.deviceFormat[0];
+ }
+ else {
+ buffer = stream_.userBuffer[0];
+ samples = stream_.bufferSize * stream_.nUserChannels[0];
+ format = stream_.userFormat;
+ }
+
+ // Do byte swapping if necessary.
+ if ( stream_.doByteSwap[0] )
+ byteSwapBuffer( buffer, samples, format );
+
+ if ( stream_.mode == DUPLEX && handle->triggered == false ) {
+ int trig = 0;
+ ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
+ result = write( handle->id[0], buffer, samples * formatBytes(format) );
+ trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT;
+ ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
+ handle->triggered = true;
+ }
+ else
+ // Write samples to device.
+ result = write( handle->id[0], buffer, samples * formatBytes(format) );
+
+ if ( result == -1 ) {
+ // We'll assume this is an underrun, though there isn't a
+ // specific means for determining that.
+ handle->xrun[0] = true;
+ errorText_ = "RtApiOss::callbackEvent: audio write error.";
+ error( RtAudioError::WARNING );
+ // Continue on to input section.
+ }
+ }
+
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+
+ // Setup parameters.
+ if ( stream_.doConvertBuffer[1] ) {
+ buffer = stream_.deviceBuffer;
+ samples = stream_.bufferSize * stream_.nDeviceChannels[1];
+ format = stream_.deviceFormat[1];
+ }
+ else {
+ buffer = stream_.userBuffer[1];
+ samples = stream_.bufferSize * stream_.nUserChannels[1];
+ format = stream_.userFormat;
+ }
+
+ // Read samples from device.
+ result = read( handle->id[1], buffer, samples * formatBytes(format) );
+
+ if ( result == -1 ) {
+ // We'll assume this is an overrun, though there isn't a
+ // specific means for determining that.
+ handle->xrun[1] = true;
+ errorText_ = "RtApiOss::callbackEvent: audio read error.";
+ error( RtAudioError::WARNING );
+ goto unlock;
+ }
+
+ // Do byte swapping if necessary.
+ if ( stream_.doByteSwap[1] )
+ byteSwapBuffer( buffer, samples, format );
+
+ // Do buffer conversion if necessary.
+ if ( stream_.doConvertBuffer[1] )
+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+ }
+
+ unlock:
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ RtApi::tickStreamTime();
+ if ( doStopStream == 1 ) this->stopStream();
+}
+
+static void *ossCallbackHandler( void *ptr )
+{
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiOss *object = (RtApiOss *) info->object;
+ bool *isRunning = &info->isRunning;
+
+ while ( *isRunning == true ) {
+ pthread_testcancel();
+ object->callbackEvent();
+ }
+
+ pthread_exit( NULL );
+}
+
+//******************** End of __LINUX_OSS__ *********************//
+#endif
+
+
+// *************************************************** //
+//
+// Protected common (OS-independent) RtAudio methods.
+//
+// *************************************************** //
+
+// This method can be modified to control the behavior of error
+// message printing.
+void RtApi :: error( RtAudioError::Type type )
+{
+ errorStream_.str(""); // clear the ostringstream
+
+ RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback;
+ if ( errorCallback ) {
+ // abortStream() can generate new error messages. Ignore them. Just keep original one.
+
+ if ( firstErrorOccurred_ )
+ return;
+
+ firstErrorOccurred_ = true;
+ const std::string errorMessage = errorText_;
+
+ if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) {
+ stream_.callbackInfo.isRunning = false; // exit from the thread
+ abortStream();
+ }
+
+ errorCallback( type, errorMessage );
+ firstErrorOccurred_ = false;
+ return;
+ }
+
+ if ( type == RtAudioError::WARNING && showWarnings_ == true )
+ std::cerr << '\n' << errorText_ << "\n\n";
+ else if ( type != RtAudioError::WARNING )
+ throw( RtAudioError( errorText_, type ) );
+}
+
+void RtApi :: verifyStream()
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApi:: a stream is not open!";
+ error( RtAudioError::INVALID_USE );
+ }
+}
+
+void RtApi :: clearStreamInfo()
+{
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
+ stream_.sampleRate = 0;
+ stream_.bufferSize = 0;
+ stream_.nBuffers = 0;
+ stream_.userFormat = 0;
+ stream_.userInterleaved = true;
+ stream_.streamTime = 0.0;
+ stream_.apiHandle = 0;
+ stream_.deviceBuffer = 0;
+ stream_.callbackInfo.callback = 0;
+ stream_.callbackInfo.userData = 0;
+ stream_.callbackInfo.isRunning = false;
+ stream_.callbackInfo.errorCallback = 0;
+ for ( int i=0; i<2; i++ ) {
+ stream_.device[i] = 11111;
+ stream_.doConvertBuffer[i] = false;
+ stream_.deviceInterleaved[i] = true;
+ stream_.doByteSwap[i] = false;
+ stream_.nUserChannels[i] = 0;
+ stream_.nDeviceChannels[i] = 0;
+ stream_.channelOffset[i] = 0;
+ stream_.deviceFormat[i] = 0;
+ stream_.latency[i] = 0;
+ stream_.userBuffer[i] = 0;
+ stream_.convertInfo[i].channels = 0;
+ stream_.convertInfo[i].inJump = 0;
+ stream_.convertInfo[i].outJump = 0;
+ stream_.convertInfo[i].inFormat = 0;
+ stream_.convertInfo[i].outFormat = 0;
+ stream_.convertInfo[i].inOffset.clear();
+ stream_.convertInfo[i].outOffset.clear();
+ }
+}
+
+unsigned int RtApi :: formatBytes( RtAudioFormat format )
+{
+ if ( format == RTAUDIO_SINT16 )
+ return 2;
+ else if ( format == RTAUDIO_SINT32 || format == RTAUDIO_FLOAT32 )
+ return 4;
+ else if ( format == RTAUDIO_FLOAT64 )
+ return 8;
+ else if ( format == RTAUDIO_SINT24 )
+ return 3;
+ else if ( format == RTAUDIO_SINT8 )
+ return 1;
+
+ errorText_ = "RtApi::formatBytes: undefined format.";
+ error( RtAudioError::WARNING );
+
+ return 0;
+}
+
+void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )
+{
+ if ( mode == INPUT ) { // convert device to user buffer
+ stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1];
+ stream_.convertInfo[mode].outJump = stream_.nUserChannels[1];
+ stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1];
+ stream_.convertInfo[mode].outFormat = stream_.userFormat;
+ }
+ else { // convert user to device buffer
+ stream_.convertInfo[mode].inJump = stream_.nUserChannels[0];
+ stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0];
+ stream_.convertInfo[mode].inFormat = stream_.userFormat;
+ stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0];
+ }
+
+ if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump )
+ stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump;
+ else
+ stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump;
+
+ // Set up the interleave/deinterleave offsets.
+ if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) {
+ if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) ||
+ ( mode == INPUT && stream_.userInterleaved ) ) {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
+ stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
+ stream_.convertInfo[mode].outOffset.push_back( k );
+ stream_.convertInfo[mode].inJump = 1;
+ }
+ }
+ else {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
+ stream_.convertInfo[mode].inOffset.push_back( k );
+ stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
+ stream_.convertInfo[mode].outJump = 1;
+ }
+ }
+ }
+ else { // no (de)interleaving
+ if ( stream_.userInterleaved ) {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
+ stream_.convertInfo[mode].inOffset.push_back( k );
+ stream_.convertInfo[mode].outOffset.push_back( k );
+ }
+ }
+ else {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
+ stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
+ stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
+ stream_.convertInfo[mode].inJump = 1;
+ stream_.convertInfo[mode].outJump = 1;
+ }
+ }
+ }
+
+ // Add channel offset.
+ if ( firstChannel > 0 ) {
+ if ( stream_.deviceInterleaved[mode] ) {
+ if ( mode == OUTPUT ) {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
+ stream_.convertInfo[mode].outOffset[k] += firstChannel;
+ }
+ else {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
+ stream_.convertInfo[mode].inOffset[k] += firstChannel;
+ }
+ }
+ else {
+ if ( mode == OUTPUT ) {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
+ stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );
+ }
+ else {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
+ stream_.convertInfo[mode].inOffset[k] += ( firstChannel * stream_.bufferSize );
+ }
+ }
+ }
+}
+
+void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, 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 == stream_.deviceBuffer && stream_.mode == DUPLEX &&
+ ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )
+ memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );
+
+ int j;
+ if (info.outFormat == RTAUDIO_FLOAT64) {
+ Float64 scale;
+ Float64 *out = (Float64 *)outBuffer;
+
+ if (info.inFormat == RTAUDIO_SINT8) {
+ signed char *in = (signed char *)inBuffer;
+ scale = 1.0 / 127.5;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT16) {
+ Int16 *in = (Int16 *)inBuffer;
+ scale = 1.0 / 32767.5;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT24) {
+ Int24 *in = (Int24 *)inBuffer;
+ scale = 1.0 / 8388607.5;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]].asInt());
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT32) {
+ Int32 *in = (Int32 *)inBuffer;
+ scale = 1.0 / 2147483647.5;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT32) {
+ Float32 *in = (Float32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT64) {
+ // Channel compensation and/or (de)interleaving only.
+ Float64 *in = (Float64 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = in[info.inOffset[j]];
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ }
+ else if (info.outFormat == RTAUDIO_FLOAT32) {
+ Float32 scale;
+ Float32 *out = (Float32 *)outBuffer;
+
+ if (info.inFormat == RTAUDIO_SINT8) {
+ signed char *in = (signed char *)inBuffer;
+ scale = (Float32) ( 1.0 / 127.5 );
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT16) {
+ Int16 *in = (Int16 *)inBuffer;
+ scale = (Float32) ( 1.0 / 32767.5 );
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT24) {
+ Int24 *in = (Int24 *)inBuffer;
+ scale = (Float32) ( 1.0 / 8388607.5 );
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]].asInt());
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT32) {
+ Int32 *in = (Int32 *)inBuffer;
+ scale = (Float32) ( 1.0 / 2147483647.5 );
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT32) {
+ // Channel compensation and/or (de)interleaving only.
+ Float32 *in = (Float32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = in[info.inOffset[j]];
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT64) {
+ Float64 *in = (Float64 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ }
+ else if (info.outFormat == RTAUDIO_SINT32) {
+ Int32 *out = (Int32 *)outBuffer;
+ if (info.inFormat == RTAUDIO_SINT8) {
+ signed char *in = (signed char *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
+ out[info.outOffset[j]] <<= 24;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT16) {
+ Int16 *in = (Int16 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
+ out[info.outOffset[j]] <<= 16;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT24) {
+ Int24 *in = (Int24 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) in[info.inOffset[j]].asInt();
+ out[info.outOffset[j]] <<= 8;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT32) {
+ // Channel compensation and/or (de)interleaving only.
+ Int32 *in = (Int32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = in[info.inOffset[j]];
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT32) {
+ Float32 *in = (Float32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT64) {
+ Float64 *in = (Float64 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ }
+ else if (info.outFormat == RTAUDIO_SINT24) {
+ Int24 *out = (Int24 *)outBuffer;
+ if (info.inFormat == RTAUDIO_SINT8) {
+ signed char *in = (signed char *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 16);
+ //out[info.outOffset[j]] <<= 16;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT16) {
+ Int16 *in = (Int16 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 8);
+ //out[info.outOffset[j]] <<= 8;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT24) {
+ // Channel compensation and/or (de)interleaving only.
+ Int24 *in = (Int24 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = in[info.inOffset[j]];
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT32) {
+ Int32 *in = (Int32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] >> 8);
+ //out[info.outOffset[j]] >>= 8;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT32) {
+ Float32 *in = (Float32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT64) {
+ Float64 *in = (Float64 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ }
+ else if (info.outFormat == RTAUDIO_SINT16) {
+ Int16 *out = (Int16 *)outBuffer;
+ if (info.inFormat == RTAUDIO_SINT8) {
+ signed char *in = (signed char *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];
+ out[info.outOffset[j]] <<= 8;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT16) {
+ // Channel compensation and/or (de)interleaving only.
+ Int16 *in = (Int16 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = in[info.inOffset[j]];
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT24) {
+ Int24 *in = (Int24 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]].asInt() >> 8);
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT32) {
+ Int32 *in = (Int32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT32) {
+ Float32 *in = (Float32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT64) {
+ Float64 *in = (Float64 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ }
+ else if (info.outFormat == RTAUDIO_SINT8) {
+ signed char *out = (signed char *)outBuffer;
+ if (info.inFormat == RTAUDIO_SINT8) {
+ // Channel compensation and/or (de)interleaving only.
+ signed char *in = (signed char *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = in[info.inOffset[j]];
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ if (info.inFormat == RTAUDIO_SINT16) {
+ Int16 *in = (Int16 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT24) {
+ Int24 *in = (Int24 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]].asInt() >> 16);
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT32) {
+ Int32 *in = (Int32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT32) {
+ Float32 *in = (Float32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT64) {
+ Float64 *in = (Float64 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ }
+}
+
+//static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }
+//static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }
+//static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }
+
+void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )
+{
+ char val;
+ char *ptr;
+
+ ptr = buffer;
+ if ( format == RTAUDIO_SINT16 ) {
+ for ( unsigned int i=0; i<samples; i++ ) {
+ // Swap 1st and 2nd bytes.
+ val = *(ptr);
+ *(ptr) = *(ptr+1);
+ *(ptr+1) = val;
+
+ // Increment 2 bytes.
+ ptr += 2;
+ }
+ }
+ else if ( format == RTAUDIO_SINT32 ||
+ format == RTAUDIO_FLOAT32 ) {
+ for ( unsigned int i=0; i<samples; i++ ) {
+ // 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 == RTAUDIO_SINT24 ) {
+ for ( unsigned int i=0; i<samples; i++ ) {
+ // Swap 1st and 3rd bytes.
+ val = *(ptr);
+ *(ptr) = *(ptr+2);
+ *(ptr+2) = val;
+
+ // Increment 2 more bytes.
+ ptr += 2;
+ }
+ }
+ else if ( format == RTAUDIO_FLOAT64 ) {
+ for ( unsigned int i=0; i<samples; i++ ) {
+ // 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;
+ }
+ }
+}
+
+ // Indentation settings for Vim and Emacs
+ //
+ // Local Variables:
+ // c-basic-offset: 2
+ // indent-tabs-mode: nil
+ // End:
+ //
+ // vim: et sts=2 sw=2
+
diff --git a/RtAudio/RtAudio.h b/RtAudio/RtAudio.h
new file mode 100644
index 0000000..11345cc
--- /dev/null
+++ b/RtAudio/RtAudio.h
@@ -0,0 +1,1163 @@
+/************************************************************************/
+/*! \class RtAudio
+ \brief Realtime audio i/o C++ classes.
+
+ RtAudio provides a common API (Application Programming Interface)
+ for realtime audio input/output across Linux (native ALSA, Jack,
+ and OSS), Macintosh OS X (CoreAudio and Jack), and Windows
+ (DirectSound, ASIO and WASAPI) operating systems.
+
+ RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
+
+ RtAudio: realtime audio i/o C++ classes
+ Copyright (c) 2001-2016 Gary P. Scavone
+
+ Permission is hereby granted, free of charge, to any person
+ obtaining a copy of this software and associated documentation files
+ (the "Software"), to deal in the Software without restriction,
+ including without limitation the rights to use, copy, modify, merge,
+ publish, distribute, sublicense, and/or sell copies of the Software,
+ and to permit persons to whom the Software is furnished to do so,
+ subject to the following conditions:
+
+ The above copyright notice and this permission notice shall be
+ included in all copies or substantial portions of the Software.
+
+ Any person wishing to distribute modifications to the Software is
+ asked to send the modifications to the original developer so that
+ they can be incorporated into the canonical version. This is,
+ however, not a binding provision of this license.
+
+ THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+ EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+ ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+ CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+ WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+*/
+/************************************************************************/
+
+/*!
+ \file RtAudio.h
+ */
+
+#ifndef __RTAUDIO_H
+#define __RTAUDIO_H
+
+#define RTAUDIO_VERSION "4.1.2"
+
+#include <string>
+#include <vector>
+#include <exception>
+#include <iostream>
+
+/*! \typedef typedef unsigned long RtAudioFormat;
+ \brief RtAudio data format type.
+
+ Support for signed integers and floats. Audio data fed to/from an
+ RtAudio stream is assumed to ALWAYS be in host byte order. The
+ internal routines will automatically take care of any necessary
+ byte-swapping between the host format and the soundcard. Thus,
+ endian-ness is not a concern in the following format definitions.
+
+ - \e RTAUDIO_SINT8: 8-bit signed integer.
+ - \e RTAUDIO_SINT16: 16-bit signed integer.
+ - \e RTAUDIO_SINT24: 24-bit signed integer.
+ - \e RTAUDIO_SINT32: 32-bit signed integer.
+ - \e RTAUDIO_FLOAT32: Normalized between plus/minus 1.0.
+ - \e RTAUDIO_FLOAT64: Normalized between plus/minus 1.0.
+*/
+typedef unsigned long RtAudioFormat;
+static const RtAudioFormat RTAUDIO_SINT8 = 0x1; // 8-bit signed integer.
+static const RtAudioFormat RTAUDIO_SINT16 = 0x2; // 16-bit signed integer.
+static const RtAudioFormat RTAUDIO_SINT24 = 0x4; // 24-bit signed integer.
+static const RtAudioFormat RTAUDIO_SINT32 = 0x8; // 32-bit signed integer.
+static const RtAudioFormat RTAUDIO_FLOAT32 = 0x10; // Normalized between plus/minus 1.0.
+static const RtAudioFormat RTAUDIO_FLOAT64 = 0x20; // Normalized between plus/minus 1.0.
+
+/*! \typedef typedef unsigned long RtAudioStreamFlags;
+ \brief RtAudio stream option flags.
+
+ The following flags can be OR'ed together to allow a client to
+ make changes to the default stream behavior:
+
+ - \e RTAUDIO_NONINTERLEAVED: Use non-interleaved buffers (default = interleaved).
+ - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency.
+ - \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use.
+ - \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA only).
+
+ By default, RtAudio streams pass and receive audio data from the
+ client in an interleaved format. By passing the
+ RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio
+ data will instead be presented in non-interleaved buffers. In
+ this case, each buffer argument in the RtAudioCallback function
+ will point to a single array of data, with \c nFrames samples for
+ each channel concatenated back-to-back. For example, the first
+ sample of data for the second channel would be located at index \c
+ nFrames (assuming the \c buffer pointer was recast to the correct
+ data type for the stream).
+
+ Certain audio APIs offer a number of parameters that influence the
+ I/O latency of a stream. By default, RtAudio will attempt to set
+ these parameters internally for robust (glitch-free) performance
+ (though some APIs, like Windows Direct Sound, make this difficult).
+ By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream()
+ function, internal stream settings will be influenced in an attempt
+ to minimize stream latency, though possibly at the expense of stream
+ performance.
+
+ If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to
+ open the input and/or output stream device(s) for exclusive use.
+ Note that this is not possible with all supported audio APIs.
+
+ If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt
+ to select realtime scheduling (round-robin) for the callback thread.
+
+ If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to
+ open the "default" PCM device when using the ALSA API. Note that this
+ will override any specified input or output device id.
+*/
+typedef unsigned int RtAudioStreamFlags;
+static const RtAudioStreamFlags RTAUDIO_NONINTERLEAVED = 0x1; // Use non-interleaved buffers (default = interleaved).
+static const RtAudioStreamFlags RTAUDIO_MINIMIZE_LATENCY = 0x2; // Attempt to set stream parameters for lowest possible latency.
+static const RtAudioStreamFlags RTAUDIO_HOG_DEVICE = 0x4; // Attempt grab device and prevent use by others.
+static const RtAudioStreamFlags RTAUDIO_SCHEDULE_REALTIME = 0x8; // Try to select realtime scheduling for callback thread.
+static const RtAudioStreamFlags RTAUDIO_ALSA_USE_DEFAULT = 0x10; // Use the "default" PCM device (ALSA only).
+
+/*! \typedef typedef unsigned long RtAudioStreamStatus;
+ \brief RtAudio stream status (over- or underflow) flags.
+
+ Notification of a stream over- or underflow is indicated by a
+ non-zero stream \c status argument in the RtAudioCallback function.
+ The stream status can be one of the following two options,
+ depending on whether the stream is open for output and/or input:
+
+ - \e RTAUDIO_INPUT_OVERFLOW: Input data was discarded because of an overflow condition at the driver.
+ - \e RTAUDIO_OUTPUT_UNDERFLOW: The output buffer ran low, likely producing a break in the output sound.
+*/
+typedef unsigned int RtAudioStreamStatus;
+static const RtAudioStreamStatus RTAUDIO_INPUT_OVERFLOW = 0x1; // Input data was discarded because of an overflow condition at the driver.
+static const RtAudioStreamStatus RTAUDIO_OUTPUT_UNDERFLOW = 0x2; // The output buffer ran low, likely causing a gap in the output sound.
+
+//! RtAudio callback function prototype.
+/*!
+ All RtAudio clients must create a function of type RtAudioCallback
+ to read and/or write data from/to the audio stream. When the
+ underlying audio system is ready for new input or output data, this
+ function will be invoked.
+
+ \param outputBuffer For output (or duplex) streams, the client
+ should write \c nFrames of audio sample frames into this
+ buffer. This argument should be recast to the datatype
+ specified when the stream was opened. For input-only
+ streams, this argument will be NULL.
+
+ \param inputBuffer For input (or duplex) streams, this buffer will
+ hold \c nFrames of input audio sample frames. This
+ argument should be recast to the datatype specified when the
+ stream was opened. For output-only streams, this argument
+ will be NULL.
+
+ \param nFrames The number of sample frames of input or output
+ data in the buffers. The actual buffer size in bytes is
+ dependent on the data type and number of channels in use.
+
+ \param streamTime The number of seconds that have elapsed since the
+ stream was started.
+
+ \param status If non-zero, this argument indicates a data overflow
+ or underflow condition for the stream. The particular
+ condition can be determined by comparison with the
+ RtAudioStreamStatus flags.
+
+ \param userData A pointer to optional data provided by the client
+ when opening the stream (default = NULL).
+
+ To continue normal stream operation, the RtAudioCallback function
+ should return a value of zero. To stop the stream and drain the
+ output buffer, the function should return a value of one. To abort
+ the stream immediately, the client should return a value of two.
+ */
+typedef int (*RtAudioCallback)( void *outputBuffer, void *inputBuffer,
+ unsigned int nFrames,
+ double streamTime,
+ RtAudioStreamStatus status,
+ void *userData );
+
+/************************************************************************/
+/*! \class RtAudioError
+ \brief Exception handling class for RtAudio.
+
+ The RtAudioError class is quite simple but it does allow errors to be
+ "caught" by RtAudioError::Type. See the RtAudio documentation to know
+ which methods can throw an RtAudioError.
+*/
+/************************************************************************/
+
+class RtAudioError : public std::exception
+{
+ public:
+ //! Defined RtAudioError types.
+ enum Type {
+ WARNING, /*!< A non-critical error. */
+ DEBUG_WARNING, /*!< A non-critical error which might be useful for debugging. */
+ UNSPECIFIED, /*!< The default, unspecified error type. */
+ NO_DEVICES_FOUND, /*!< No devices found on system. */
+ INVALID_DEVICE, /*!< An invalid device ID was specified. */
+ MEMORY_ERROR, /*!< An error occured during memory allocation. */
+ INVALID_PARAMETER, /*!< An invalid parameter was specified to a function. */
+ INVALID_USE, /*!< The function was called incorrectly. */
+ DRIVER_ERROR, /*!< A system driver error occured. */
+ SYSTEM_ERROR, /*!< A system error occured. */
+ THREAD_ERROR /*!< A thread error occured. */
+ };
+
+ //! The constructor.
+ RtAudioError( const std::string& message, Type type = RtAudioError::UNSPECIFIED ) throw() : message_(message), type_(type) {}
+
+ //! The destructor.
+ virtual ~RtAudioError( void ) throw() {}
+
+ //! Prints thrown error message to stderr.
+ virtual void printMessage( void ) const throw() { std::cerr << '\n' << message_ << "\n\n"; }
+
+ //! Returns the thrown error message type.
+ virtual const Type& getType(void) const throw() { return type_; }
+
+ //! Returns the thrown error message string.
+ virtual const std::string& getMessage(void) const throw() { return message_; }
+
+ //! Returns the thrown error message as a c-style string.
+ virtual const char* what( void ) const throw() { return message_.c_str(); }
+
+ protected:
+ std::string message_;
+ Type type_;
+};
+
+//! RtAudio error callback function prototype.
+/*!
+ \param type Type of error.
+ \param errorText Error description.
+ */
+typedef void (*RtAudioErrorCallback)( RtAudioError::Type type, const std::string &errorText );
+
+// **************************************************************** //
+//
+// RtAudio class declaration.
+//
+// RtAudio 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 RtApi;
+
+class RtAudio
+{
+ public:
+
+ //! Audio API specifier arguments.
+ enum Api {
+ UNSPECIFIED, /*!< Search for a working compiled API. */
+ LINUX_ALSA, /*!< The Advanced Linux Sound Architecture API. */
+ LINUX_PULSE, /*!< The Linux PulseAudio API. */
+ LINUX_OSS, /*!< The Linux Open Sound System API. */
+ UNIX_JACK, /*!< The Jack Low-Latency Audio Server API. */
+ MACOSX_CORE, /*!< Macintosh OS-X Core Audio API. */
+ WINDOWS_WASAPI, /*!< The Microsoft WASAPI API. */
+ WINDOWS_ASIO, /*!< The Steinberg Audio Stream I/O API. */
+ WINDOWS_DS, /*!< The Microsoft Direct Sound API. */
+ RTAUDIO_DUMMY /*!< A compilable but non-functional API. */
+ };
+
+ //! The public device information structure for returning queried values.
+ struct DeviceInfo {
+ bool probed; /*!< true if the device capabilities were successfully probed. */
+ std::string name; /*!< Character string device identifier. */
+ unsigned int outputChannels; /*!< Maximum output channels supported by device. */
+ unsigned int inputChannels; /*!< Maximum input channels supported by device. */
+ unsigned int 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::vector<unsigned int> sampleRates; /*!< Supported sample rates (queried from list of standard rates). */
+ unsigned int preferredSampleRate; /*!< Preferred sample rate, eg. for WASAPI the system sample rate. */
+ RtAudioFormat nativeFormats; /*!< Bit mask of supported data formats. */
+
+ // Default constructor.
+ DeviceInfo()
+ :probed(false), outputChannels(0), inputChannels(0), duplexChannels(0),
+ isDefaultOutput(false), isDefaultInput(false), preferredSampleRate(0), nativeFormats(0) {}
+ };
+
+ //! The structure for specifying input or ouput stream parameters.
+ struct StreamParameters {
+ unsigned int deviceId; /*!< Device index (0 to getDeviceCount() - 1). */
+ unsigned int nChannels; /*!< Number of channels. */
+ unsigned int firstChannel; /*!< First channel index on device (default = 0). */
+
+ // Default constructor.
+ StreamParameters()
+ : deviceId(0), nChannels(0), firstChannel(0) {}
+ };
+
+ //! The structure for specifying stream options.
+ /*!
+ The following flags can be OR'ed together to allow a client to
+ make changes to the default stream behavior:
+
+ - \e RTAUDIO_NONINTERLEAVED: Use non-interleaved buffers (default = interleaved).
+ - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency.
+ - \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use.
+ - \e RTAUDIO_SCHEDULE_REALTIME: Attempt to select realtime scheduling for callback thread.
+ - \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA only).
+
+ By default, RtAudio streams pass and receive audio data from the
+ client in an interleaved format. By passing the
+ RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio
+ data will instead be presented in non-interleaved buffers. In
+ this case, each buffer argument in the RtAudioCallback function
+ will point to a single array of data, with \c nFrames samples for
+ each channel concatenated back-to-back. For example, the first
+ sample of data for the second channel would be located at index \c
+ nFrames (assuming the \c buffer pointer was recast to the correct
+ data type for the stream).
+
+ Certain audio APIs offer a number of parameters that influence the
+ I/O latency of a stream. By default, RtAudio will attempt to set
+ these parameters internally for robust (glitch-free) performance
+ (though some APIs, like Windows Direct Sound, make this difficult).
+ By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream()
+ function, internal stream settings will be influenced in an attempt
+ to minimize stream latency, though possibly at the expense of stream
+ performance.
+
+ If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to
+ open the input and/or output stream device(s) for exclusive use.
+ Note that this is not possible with all supported audio APIs.
+
+ If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt
+ to select realtime scheduling (round-robin) for the callback thread.
+ The \c priority parameter will only be used if the RTAUDIO_SCHEDULE_REALTIME
+ flag is set. It defines the thread's realtime priority.
+
+ If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to
+ open the "default" PCM device when using the ALSA API. Note that this
+ will override any specified input or output device id.
+
+ The \c numberOfBuffers parameter can be used to control stream
+ latency in the Windows DirectSound, Linux OSS, and Linux Alsa APIs
+ only. A value of two is usually the smallest allowed. Larger
+ numbers can potentially result in more robust stream performance,
+ though likely at the cost of stream latency. The value set by the
+ user is replaced during execution of the RtAudio::openStream()
+ function by the value actually used by the system.
+
+ The \c streamName parameter can be used to set the client name
+ when using the Jack API. By default, the client name is set to
+ RtApiJack. However, if you wish to create multiple instances of
+ RtAudio with Jack, each instance must have a unique client name.
+ */
+ struct StreamOptions {
+ RtAudioStreamFlags flags; /*!< A bit-mask of stream flags (RTAUDIO_NONINTERLEAVED, RTAUDIO_MINIMIZE_LATENCY, RTAUDIO_HOG_DEVICE, RTAUDIO_ALSA_USE_DEFAULT). */
+ unsigned int numberOfBuffers; /*!< Number of stream buffers. */
+ std::string streamName; /*!< A stream name (currently used only in Jack). */
+ int priority; /*!< Scheduling priority of callback thread (only used with flag RTAUDIO_SCHEDULE_REALTIME). */
+
+ // Default constructor.
+ StreamOptions()
+ : flags(0), numberOfBuffers(0), priority(0) {}
+ };
+
+ //! A static function to determine the current RtAudio version.
+ static std::string getVersion( void ) throw();
+
+ //! A static function to determine the available compiled audio APIs.
+ /*!
+ The values returned in the std::vector can be compared against
+ the enumerated list values. Note that there can be more than one
+ API compiled for certain operating systems.
+ */
+ static void getCompiledApi( std::vector<RtAudio::Api> &apis ) throw();
+
+ //! The class constructor.
+ /*!
+ The constructor performs minor initialization tasks. An exception
+ can be thrown if no API support is compiled.
+
+ If no API argument is specified and multiple API support has been
+ compiled, the default order of use is JACK, ALSA, OSS (Linux
+ systems) and ASIO, DS (Windows systems).
+ */
+ RtAudio( RtAudio::Api api=UNSPECIFIED );
+
+ //! The destructor.
+ /*!
+ If a stream is running or open, it will be stopped and closed
+ automatically.
+ */
+ ~RtAudio() throw();
+
+ //! Returns the audio API specifier for the current instance of RtAudio.
+ RtAudio::Api getCurrentApi( void ) throw();
+
+ //! 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.
+ */
+ unsigned int getDeviceCount( void ) throw();
+
+ //! Return an RtAudio::DeviceInfo structure for a specified device number.
+ /*!
+
+ Any device integer between 0 and getDeviceCount() - 1 is valid.
+ If an invalid argument is provided, an RtAudioError (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".
+ */
+ RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
+
+ //! 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.
+ */
+ unsigned int getDefaultOutputDevice( void ) throw();
+
+ //! 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.
+ */
+ unsigned int getDefaultInputDevice( void ) throw();
+
+ //! A public function for opening a stream with the specified parameters.
+ /*!
+ An RtAudioError (type = SYSTEM_ERROR) is thrown if a stream cannot be
+ opened with the specified parameters or an error occurs during
+ processing. An RtAudioError (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 RtAudioFormat 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 userData An optional pointer to data that can be accessed
+ from within the callback function.
+ \param options An optional pointer to a structure containing various
+ global stream options, including a list of OR'ed RtAudioStreamFlags
+ 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 RTAUDIO_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.
+ */
+ void openStream( RtAudio::StreamParameters *outputParameters,
+ RtAudio::StreamParameters *inputParameters,
+ RtAudioFormat format, unsigned int sampleRate,
+ unsigned int *bufferFrames, RtAudioCallback callback,
+ void *userData = NULL, RtAudio::StreamOptions *options = NULL, RtAudioErrorCallback errorCallback = NULL );
+
+ //! 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).
+ */
+ void closeStream( void ) throw();
+
+ //! A function that starts a stream.
+ /*!
+ An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs
+ during processing. An RtAudioError (type = INVALID_USE) is thrown if a
+ stream is not open. A warning is issued if the stream is already
+ running.
+ */
+ void startStream( void );
+
+ //! Stop a stream, allowing any samples remaining in the output queue to be played.
+ /*!
+ An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs
+ during processing. An RtAudioError (type = INVALID_USE) is thrown if a
+ stream is not open. A warning is issued if the stream is already
+ stopped.
+ */
+ void stopStream( void );
+
+ //! Stop a stream, discarding any samples remaining in the input/output queue.
+ /*!
+ An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs
+ during processing. An RtAudioError (type = INVALID_USE) is thrown if a
+ stream is not open. A warning is issued if the stream is already
+ stopped.
+ */
+ void abortStream( void );
+
+ //! Returns true if a stream is open and false if not.
+ bool isStreamOpen( void ) const throw();
+
+ //! Returns true if the stream is running and false if it is stopped or not open.
+ bool isStreamRunning( void ) const throw();
+
+ //! Returns the number of elapsed seconds since the stream was started.
+ /*!
+ If a stream is not open, an RtAudioError (type = INVALID_USE) will be thrown.
+ */
+ double getStreamTime( void );
+
+ //! Set the stream time to a time in seconds greater than or equal to 0.0.
+ /*!
+ If a stream is not open, an RtAudioError (type = INVALID_USE) will be thrown.
+ */
+ void setStreamTime( double time );
+
+ //! Returns the internal stream latency in sample frames.
+ /*!
+ 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
+ RtAudioError (type = INVALID_USE) will be thrown. If the API does not
+ report latency, the return value will be zero.
+ */
+ long getStreamLatency( void );
+
+ //! Returns actual sample rate in use by the stream.
+ /*!
+ 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 RtAudioError (type = INVALID_USE) will be thrown.
+ */
+ unsigned int getStreamSampleRate( void );
+
+ //! Specify whether warning messages should be printed to stderr.
+ void showWarnings( bool value = true ) throw();
+
+ protected:
+
+ void openRtApi( RtAudio::Api api );
+ RtApi *rtapi_;
+};
+
+// Operating system dependent thread functionality.
+#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__)
+
+ #ifndef NOMINMAX
+ #define NOMINMAX
+ #endif
+ #include <windows.h>
+ #include <process.h>
+
+ typedef uintptr_t ThreadHandle;
+ typedef CRITICAL_SECTION StreamMutex;
+
+#elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
+ // Using pthread library for various flavors of unix.
+ #include <pthread.h>
+
+ typedef pthread_t ThreadHandle;
+ typedef pthread_mutex_t StreamMutex;
+
+#else // Setup for "dummy" behavior
+
+ #define __RTAUDIO_DUMMY__
+ typedef int ThreadHandle;
+ typedef int StreamMutex;
+
+#endif
+
+// This global structure type is used to pass callback information
+// between the private RtAudio stream structure and global callback
+// handling functions.
+struct CallbackInfo {
+ void *object; // Used as a "this" pointer.
+ ThreadHandle thread;
+ void *callback;
+ void *userData;
+ void *errorCallback;
+ void *apiInfo; // void pointer for API specific callback information
+ bool isRunning;
+ bool doRealtime;
+ int priority;
+
+ // Default constructor.
+ CallbackInfo()
+ :object(0), callback(0), userData(0), errorCallback(0), apiInfo(0), isRunning(false), doRealtime(false) {}
+};
+
+// **************************************************************** //
+//
+// RtApi class declaration.
+//
+// Subclasses of RtApi contain all API- and OS-specific code necessary
+// to fully implement the RtAudio API.
+//
+// Note that RtApi is an abstract base class and cannot be
+// explicitly instantiated. The class RtAudio will create an
+// instance of an RtApi subclass (RtApiOss, RtApiAlsa,
+// RtApiJack, RtApiCore, RtApiDs, or RtApiAsio).
+//
+// **************************************************************** //
+
+#pragma pack(push, 1)
+class S24 {
+
+ protected:
+ unsigned char c3[3];
+
+ public:
+ S24() {}
+
+ S24& operator = ( const int& i ) {
+ c3[0] = (i & 0x000000ff);
+ c3[1] = (i & 0x0000ff00) >> 8;
+ c3[2] = (i & 0x00ff0000) >> 16;
+ return *this;
+ }
+
+ S24( const S24& v ) { *this = v; }
+ S24( const double& d ) { *this = (int) d; }
+ S24( const float& f ) { *this = (int) f; }
+ S24( const signed short& s ) { *this = (int) s; }
+ S24( const char& c ) { *this = (int) c; }
+
+ int asInt() {
+ int i = c3[0] | (c3[1] << 8) | (c3[2] << 16);
+ if (i & 0x800000) i |= ~0xffffff;
+ return i;
+ }
+};
+#pragma pack(pop)
+
+#if defined( HAVE_GETTIMEOFDAY )
+ #include <sys/time.h>
+#endif
+
+#include <sstream>
+
+class RtApi
+{
+public:
+
+ RtApi();
+ virtual ~RtApi();
+ virtual RtAudio::Api getCurrentApi( void ) = 0;
+ virtual unsigned int getDeviceCount( void ) = 0;
+ virtual RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) = 0;
+ virtual unsigned int getDefaultInputDevice( void );
+ virtual unsigned int getDefaultOutputDevice( void );
+ void openStream( RtAudio::StreamParameters *outputParameters,
+ RtAudio::StreamParameters *inputParameters,
+ RtAudioFormat format, unsigned int sampleRate,
+ unsigned int *bufferFrames, RtAudioCallback callback,
+ void *userData, RtAudio::StreamOptions *options,
+ RtAudioErrorCallback errorCallback );
+ virtual void closeStream( void );
+ virtual void startStream( void ) = 0;
+ virtual void stopStream( void ) = 0;
+ virtual void abortStream( void ) = 0;
+ long getStreamLatency( void );
+ unsigned int getStreamSampleRate( void );
+ virtual double getStreamTime( void );
+ virtual void setStreamTime( double time );
+ bool isStreamOpen( void ) const { return stream_.state != STREAM_CLOSED; }
+ bool isStreamRunning( void ) const { return stream_.state == STREAM_RUNNING; }
+ void showWarnings( bool value ) { showWarnings_ = value; }
+
+
+protected:
+
+ static const unsigned int MAX_SAMPLE_RATES;
+ static const unsigned int SAMPLE_RATES[];
+
+ enum { FAILURE, SUCCESS };
+
+ enum StreamState {
+ STREAM_STOPPED,
+ STREAM_STOPPING,
+ STREAM_RUNNING,
+ STREAM_CLOSED = -50
+ };
+
+ enum StreamMode {
+ OUTPUT,
+ INPUT,
+ DUPLEX,
+ UNINITIALIZED = -75
+ };
+
+ // A protected structure used for buffer conversion.
+ struct ConvertInfo {
+ int channels;
+ int inJump, outJump;
+ RtAudioFormat inFormat, outFormat;
+ std::vector<int> inOffset;
+ std::vector<int> outOffset;
+ };
+
+ // A protected structure for audio streams.
+ struct RtApiStream {
+ unsigned int device[2]; // Playback and record, respectively.
+ void *apiHandle; // void pointer for API specific stream handle information
+ StreamMode mode; // OUTPUT, INPUT, or DUPLEX.
+ StreamState state; // STOPPED, RUNNING, or CLOSED
+ char *userBuffer[2]; // Playback and record, respectively.
+ char *deviceBuffer;
+ bool doConvertBuffer[2]; // Playback and record, respectively.
+ bool userInterleaved;
+ bool deviceInterleaved[2]; // Playback and record, respectively.
+ bool doByteSwap[2]; // Playback and record, respectively.
+ unsigned int sampleRate;
+ unsigned int bufferSize;
+ unsigned int nBuffers;
+ unsigned int nUserChannels[2]; // Playback and record, respectively.
+ unsigned int nDeviceChannels[2]; // Playback and record channels, respectively.
+ unsigned int channelOffset[2]; // Playback and record, respectively.
+ unsigned long latency[2]; // Playback and record, respectively.
+ RtAudioFormat userFormat;
+ RtAudioFormat deviceFormat[2]; // Playback and record, respectively.
+ StreamMutex mutex;
+ CallbackInfo callbackInfo;
+ ConvertInfo convertInfo[2];
+ double streamTime; // Number of elapsed seconds since the stream started.
+
+#if defined(HAVE_GETTIMEOFDAY)
+ struct timeval lastTickTimestamp;
+#endif
+
+ RtApiStream()
+ :apiHandle(0), deviceBuffer(0) { device[0] = 11111; device[1] = 11111; }
+ };
+
+ typedef S24 Int24;
+ typedef signed short Int16;
+ typedef signed int Int32;
+ typedef float Float32;
+ typedef double Float64;
+
+ std::ostringstream errorStream_;
+ std::string errorText_;
+ bool showWarnings_;
+ RtApiStream stream_;
+ bool firstErrorOccurred_;
+
+ /*!
+ Protected, 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 FAILURE is returned. A
+ successful probe is indicated by a return value of SUCCESS.
+ */
+ virtual bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options );
+
+ //! A protected function used to increment the stream time.
+ void tickStreamTime( void );
+
+ //! Protected common method to clear an RtApiStream structure.
+ void clearStreamInfo();
+
+ /*!
+ Protected common method that throws an RtAudioError (type =
+ INVALID_USE) if a stream is not open.
+ */
+ void verifyStream( void );
+
+ //! Protected common error method to allow global control over error handling.
+ void error( RtAudioError::Type type );
+
+ /*!
+ Protected method used to perform format, channel number, and/or interleaving
+ conversions between the user and device buffers.
+ */
+ void convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info );
+
+ //! Protected common method used to perform byte-swapping on buffers.
+ void byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format );
+
+ //! Protected common method that returns the number of bytes for a given format.
+ unsigned int formatBytes( RtAudioFormat format );
+
+ //! Protected common method that sets up the parameters for buffer conversion.
+ void setConvertInfo( StreamMode mode, unsigned int firstChannel );
+};
+
+// **************************************************************** //
+//
+// Inline RtAudio definitions.
+//
+// **************************************************************** //
+
+inline RtAudio::Api RtAudio :: getCurrentApi( void ) throw() { return rtapi_->getCurrentApi(); }
+inline unsigned int RtAudio :: getDeviceCount( void ) throw() { return rtapi_->getDeviceCount(); }
+inline RtAudio::DeviceInfo RtAudio :: getDeviceInfo( unsigned int device ) { return rtapi_->getDeviceInfo( device ); }
+inline unsigned int RtAudio :: getDefaultInputDevice( void ) throw() { return rtapi_->getDefaultInputDevice(); }
+inline unsigned int RtAudio :: getDefaultOutputDevice( void ) throw() { return rtapi_->getDefaultOutputDevice(); }
+inline void RtAudio :: closeStream( void ) throw() { return rtapi_->closeStream(); }
+inline void RtAudio :: startStream( void ) { return rtapi_->startStream(); }
+inline void RtAudio :: stopStream( void ) { return rtapi_->stopStream(); }
+inline void RtAudio :: abortStream( void ) { return rtapi_->abortStream(); }
+inline bool RtAudio :: isStreamOpen( void ) const throw() { return rtapi_->isStreamOpen(); }
+inline bool RtAudio :: isStreamRunning( void ) const throw() { return rtapi_->isStreamRunning(); }
+inline long RtAudio :: getStreamLatency( void ) { return rtapi_->getStreamLatency(); }
+inline unsigned int RtAudio :: getStreamSampleRate( void ) { return rtapi_->getStreamSampleRate(); }
+inline double RtAudio :: getStreamTime( void ) { return rtapi_->getStreamTime(); }
+inline void RtAudio :: setStreamTime( double time ) { return rtapi_->setStreamTime( time ); }
+inline void RtAudio :: showWarnings( bool value ) throw() { rtapi_->showWarnings( value ); }
+
+// RtApi Subclass prototypes.
+
+#if defined(__MACOSX_CORE__)
+
+#include <CoreAudio/AudioHardware.h>
+
+class RtApiCore: public RtApi
+{
+public:
+
+ RtApiCore();
+ ~RtApiCore();
+ RtAudio::Api getCurrentApi( void ) { return RtAudio::MACOSX_CORE; }
+ unsigned int getDeviceCount( void );
+ RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
+ unsigned int getDefaultOutputDevice( void );
+ unsigned int getDefaultInputDevice( void );
+ void closeStream( void );
+ void startStream( void );
+ void stopStream( void );
+ void abortStream( void );
+ long getStreamLatency( void );
+
+ // 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( AudioDeviceID deviceId,
+ const AudioBufferList *inBufferList,
+ const AudioBufferList *outBufferList );
+
+ private:
+
+ bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options );
+ static const char* getErrorCode( OSStatus code );
+};
+
+#endif
+
+#if defined(__UNIX_JACK__)
+
+class RtApiJack: public RtApi
+{
+public:
+
+ RtApiJack();
+ ~RtApiJack();
+ RtAudio::Api getCurrentApi( void ) { return RtAudio::UNIX_JACK; }
+ unsigned int getDeviceCount( void );
+ RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
+ void closeStream( void );
+ void startStream( void );
+ void stopStream( void );
+ void abortStream( void );
+ long getStreamLatency( void );
+
+ // 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( unsigned long nframes );
+
+ private:
+
+ bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options );
+};
+
+#endif
+
+#if defined(__WINDOWS_ASIO__)
+
+class RtApiAsio: public RtApi
+{
+public:
+
+ RtApiAsio();
+ ~RtApiAsio();
+ RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_ASIO; }
+ unsigned int getDeviceCount( void );
+ RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
+ void closeStream( void );
+ void startStream( void );
+ void stopStream( void );
+ void abortStream( void );
+ long getStreamLatency( void );
+
+ // 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:
+
+ std::vector<RtAudio::DeviceInfo> devices_;
+ void saveDeviceInfo( void );
+ bool coInitialized_;
+ bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options );
+};
+
+#endif
+
+#if defined(__WINDOWS_DS__)
+
+class RtApiDs: public RtApi
+{
+public:
+
+ RtApiDs();
+ ~RtApiDs();
+ RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_DS; }
+ unsigned int getDeviceCount( void );
+ unsigned int getDefaultOutputDevice( void );
+ unsigned int getDefaultInputDevice( void );
+ RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
+ void closeStream( void );
+ void startStream( void );
+ void stopStream( void );
+ void abortStream( void );
+ long getStreamLatency( void );
+
+ // 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 );
+
+ private:
+
+ bool coInitialized_;
+ bool buffersRolling;
+ long duplexPrerollBytes;
+ std::vector<struct DsDevice> dsDevices;
+ bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options );
+};
+
+#endif
+
+#if defined(__WINDOWS_WASAPI__)
+
+struct IMMDeviceEnumerator;
+
+class RtApiWasapi : public RtApi
+{
+public:
+ RtApiWasapi();
+ ~RtApiWasapi();
+
+ RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_WASAPI; }
+ unsigned int getDeviceCount( void );
+ RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
+ unsigned int getDefaultOutputDevice( void );
+ unsigned int getDefaultInputDevice( void );
+ void closeStream( void );
+ void startStream( void );
+ void stopStream( void );
+ void abortStream( void );
+
+private:
+ bool coInitialized_;
+ IMMDeviceEnumerator* deviceEnumerator_;
+
+ bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int* bufferSize,
+ RtAudio::StreamOptions* options );
+
+ static DWORD WINAPI runWasapiThread( void* wasapiPtr );
+ static DWORD WINAPI stopWasapiThread( void* wasapiPtr );
+ static DWORD WINAPI abortWasapiThread( void* wasapiPtr );
+ void wasapiThread();
+};
+
+#endif
+
+#if defined(__LINUX_ALSA__)
+
+class RtApiAlsa: public RtApi
+{
+public:
+
+ RtApiAlsa();
+ ~RtApiAlsa();
+ RtAudio::Api getCurrentApi() { return RtAudio::LINUX_ALSA; }
+ unsigned int getDeviceCount( void );
+ RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
+ void closeStream( void );
+ void startStream( void );
+ void stopStream( void );
+ void abortStream( void );
+
+ // 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 );
+
+ private:
+
+ std::vector<RtAudio::DeviceInfo> devices_;
+ void saveDeviceInfo( void );
+ bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options );
+};
+
+#endif
+
+#if defined(__LINUX_PULSE__)
+
+class RtApiPulse: public RtApi
+{
+public:
+ ~RtApiPulse();
+ RtAudio::Api getCurrentApi() { return RtAudio::LINUX_PULSE; }
+ unsigned int getDeviceCount( void );
+ RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
+ void closeStream( void );
+ void startStream( void );
+ void stopStream( void );
+ void abortStream( void );
+
+ // 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 );
+
+ private:
+
+ std::vector<RtAudio::DeviceInfo> devices_;
+ void saveDeviceInfo( void );
+ bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options );
+};
+
+#endif
+
+#if defined(__LINUX_OSS__)
+
+class RtApiOss: public RtApi
+{
+public:
+
+ RtApiOss();
+ ~RtApiOss();
+ RtAudio::Api getCurrentApi() { return RtAudio::LINUX_OSS; }
+ unsigned int getDeviceCount( void );
+ RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
+ void closeStream( void );
+ void startStream( void );
+ void stopStream( void );
+ void abortStream( void );
+
+ // 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 );
+
+ private:
+
+ bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options );
+};
+
+#endif
+
+#if defined(__RTAUDIO_DUMMY__)
+
+class RtApiDummy: public RtApi
+{
+public:
+
+ RtApiDummy() { errorText_ = "RtApiDummy: This class provides no functionality."; error( RtAudioError::WARNING ); }
+ RtAudio::Api getCurrentApi( void ) { return RtAudio::RTAUDIO_DUMMY; }
+ unsigned int getDeviceCount( void ) { return 0; }
+ RtAudio::DeviceInfo getDeviceInfo( unsigned int /*device*/ ) { RtAudio::DeviceInfo info; return info; }
+ void closeStream( void ) {}
+ void startStream( void ) {}
+ void stopStream( void ) {}
+ void abortStream( void ) {}
+
+ private:
+
+ bool probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/,
+ unsigned int /*firstChannel*/, unsigned int /*sampleRate*/,
+ RtAudioFormat /*format*/, unsigned int * /*bufferSize*/,
+ RtAudio::StreamOptions * /*options*/ ) { return false; }
+};
+
+#endif
+
+#endif
+
+// Indentation settings for Vim and Emacs
+//
+// Local Variables:
+// c-basic-offset: 2
+// indent-tabs-mode: nil
+// End:
+//
+// vim: et sts=2 sw=2
diff --git a/RtAudio/readme b/RtAudio/readme
new file mode 100644
index 0000000..079875f
--- /dev/null
+++ b/RtAudio/readme
@@ -0,0 +1,61 @@
+RtAudio - a set of C++ classes that provide a common API for realtime audio input/output across Linux (native ALSA, JACK, PulseAudio and OSS), Macintosh OS X (CoreAudio and JACK), and Windows (DirectSound, ASIO and WASAPI) operating systems.
+
+By Gary P. Scavone, 2001-2016.
+
+This distribution of RtAudio contains the following:
+
+doc: RtAudio documentation (see doc/html/index.html)
+tests: example RtAudio programs
+include: header and source files necessary for ASIO, DS & OSS compilation
+tests/Windows: Visual C++ .net test program workspace and projects
+
+OVERVIEW:
+
+RtAudio is a set of C++ classes that provides a common API (Application Programming Interface) for realtime audio input/output across Linux (native ALSA, JACK, PulseAudio and OSS), Macintosh OS X and Windows (DirectSound, ASIO and WASAPI) operating systems. RtAudio significantly simplifies the process of interacting with computer audio hardware. It was designed with the following objectives:
+
+ - object-oriented C++ design
+ - simple, common API across all supported platforms
+ - only one source and one header file for easy inclusion in programming projects
+ - allow simultaneous multi-api support
+ - support dynamic connection of devices
+ - provide extensive audio device parameter control
+ - allow audio device capability probing
+ - automatic internal conversion for data format, channel number compensation, (de)interleaving, and byte-swapping
+
+RtAudio incorporates the concept of audio streams, which represent audio output (playback) and/or input (recording). Available audio devices and their capabilities can be enumerated and then specified when opening a stream. Where applicable, multiple API support can be compiled and a particular API specified when creating an RtAudio instance. See the \ref apinotes section for information specific to each of the supported audio APIs.
+
+FURTHER READING:
+
+For complete documentation on RtAudio, see the doc directory of the distribution or surf to http://www.music.mcgill.ca/~gary/rtaudio/.
+
+
+LEGAL AND ETHICAL:
+
+The RtAudio license is similar to the MIT License.
+
+ RtAudio: a set of realtime audio i/o C++ classes
+ Copyright (c) 2001-2016 Gary P. Scavone
+
+ Permission is hereby granted, free of charge, to any person
+ obtaining a copy of this software and associated documentation files
+ (the "Software"), to deal in the Software without restriction,
+ including without limitation the rights to use, copy, modify, merge,
+ publish, distribute, sublicense, and/or sell copies of the Software,
+ and to permit persons to whom the Software is furnished to do so,
+ subject to the following conditions:
+
+ The above copyright notice and this permission notice shall be
+ included in all copies or substantial portions of the Software.
+
+ Any person wishing to distribute modifications to the Software is
+ asked to send the modifications to the original developer so that
+ they can be incorporated into the canonical version. This is,
+ however, not a binding provision of this license.
+
+ THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+ EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+ ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+ CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+ WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
diff --git a/Settings.cpp b/Settings.cpp
new file mode 100644
index 0000000..172e5e4
--- /dev/null
+++ b/Settings.cpp
@@ -0,0 +1,603 @@
+/*
+ * The MIT License (MIT)
+ *
+ * Copyright (c) 2015 Charles J. Cliffe
+
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+#include "SoapyAudio.hpp"
+
+#ifdef USE_HAMLIB
+std::vector<const struct rig_caps *> SoapyAudio::rigCaps;
+#endif
+
+SoapyAudio::SoapyAudio(const SoapySDR::Kwargs &args)
+{
+ deviceId = -1;
+
+ asFormat = AUDIO_FORMAT_FLOAT32;
+
+ sampleRate = 48000;
+ centerFrequency = 0;
+
+ numBuffers = DEFAULT_NUM_BUFFERS;
+
+ agcMode = false;
+
+ bufferedElems = 0;
+ resetBuffer = false;
+
+ streamActive = false;
+ sampleRateChanged.store(false);
+
+ sampleOffset = 0;
+
+ if (args.count("device_id") != 0)
+ {
+ try {
+ deviceId = std::stoi(args.at("device_id"));
+ } catch (const std::invalid_argument &) {
+ }
+
+ int numDevices = dac.getDeviceCount();
+
+ if (deviceId < 0 || deviceId >= numDevices)
+ {
+ throw std::runtime_error(
+ "device_id out of range [0 .. " + std::to_string(numDevices) + "].");
+ }
+
+ SoapySDR_logf(SOAPY_SDR_DEBUG, "Found Audio device using 'device_id' = %d", deviceId);
+ }
+
+ if (deviceId == -1) {
+ throw std::runtime_error("device_id missing.");
+ }
+
+ RtAudio endac;
+
+ devInfo = endac.getDeviceInfo(deviceId);
+
+#ifdef USE_HAMLIB
+ t_Rig = nullptr;
+ rigThread = nullptr;
+ rigModel = 0;
+ rigFile = "";
+ rigSerialRate = 0;
+
+ if (args.count("rig") != 0 && args.at("rig") != "") {
+ try {
+ rigModel = std::stoi(args.at("rig"));
+ } catch (const std::invalid_argument &) {
+ throw std::runtime_error("rig is invalid.");
+ }
+ if (!args.count("rig_rate")) {
+ throw std::runtime_error("rig_rate missing.");
+ }
+ try {
+ rigSerialRate = std::stoi(args.at("rig_rate"));
+ } catch (const std::invalid_argument &) {
+ throw std::runtime_error("rig_rate is invalid.");
+ }
+
+ if (!args.count("rig_port")) {
+ throw std::runtime_error("rig_port missing.");
+ }
+ rigFile = args.at("rig_port");
+ checkRigThread();
+ }
+#endif
+}
+
+SoapyAudio::~SoapyAudio(void)
+{
+#ifdef USE_HAMLIB
+ if (rigThread) {
+ if (!rigThread->isTerminated()) {
+ rigThread->terminate();
+ }
+ if (t_Rig && t_Rig->joinable()) {
+ t_Rig->join();
+ }
+ }
+#endif
+}
+
+/*******************************************************************
+ * Identification API
+ ******************************************************************/
+
+std::string SoapyAudio::getDriverKey(void) const
+{
+ return "Audio";
+}
+
+std::string SoapyAudio::getHardwareKey(void) const
+{
+ return "Audio";
+}
+
+SoapySDR::Kwargs SoapyAudio::getHardwareInfo(void) const
+{
+ //key/value pairs for any useful information
+ //this also gets printed in --probe
+ SoapySDR::Kwargs args;
+
+ args["origin"] = "https://github.com/pothosware/SoapyAudio";
+ args["device_id"] = std::to_string(deviceId);
+
+ return args;
+}
+
+/*******************************************************************
+ * Channels API
+ ******************************************************************/
+
+size_t SoapyAudio::getNumChannels(const int dir) const
+{
+ return (dir == SOAPY_SDR_RX) ? 1 : 0;
+}
+
+/*******************************************************************
+ * Antenna API
+ ******************************************************************/
+
+std::vector<std::string> SoapyAudio::listAntennas(const int direction, const size_t channel) const
+{
+ std::vector<std::string> antennas;
+ antennas.push_back("RX");
+ // antennas.push_back("TX");
+ return antennas;
+}
+
+void SoapyAudio::setAntenna(const int direction, const size_t channel, const std::string &name)
+{
+ // TODO
+}
+
+std::string SoapyAudio::getAntenna(const int direction, const size_t channel) const
+{
+ return "RX";
+ // return "TX";
+}
+
+/*******************************************************************
+ * Frontend corrections API
+ ******************************************************************/
+
+bool SoapyAudio::hasDCOffsetMode(const int direction, const size_t channel) const
+{
+ return false;
+}
+
+/*******************************************************************
+ * Gain API
+ ******************************************************************/
+
+std::vector<std::string> SoapyAudio::listGains(const int direction, const size_t channel) const
+{
+ //list available gain elements,
+ //the functions below have a "name" parameter
+ std::vector<std::string> results;
+
+ // results.push_back("AUDIO");
+
+ return results;
+}
+
+bool SoapyAudio::hasGainMode(const int direction, const size_t channel) const
+{
+ return true;
+}
+
+void SoapyAudio::setGainMode(const int direction, const size_t channel, const bool automatic)
+{
+ agcMode = automatic;
+ SoapySDR_logf(SOAPY_SDR_DEBUG, "Setting Audio AGC: %s", automatic ? "Automatic" : "Manual");
+}
+
+bool SoapyAudio::getGainMode(const int direction, const size_t channel) const
+{
+ return agcMode;
+}
+
+void SoapyAudio::setGain(const int direction, const size_t channel, const double value)
+{
+ //set the overall gain by distributing it across available gain elements
+ //OR delete this function to use SoapySDR's default gain distribution algorithm...
+ SoapySDR::Device::setGain(direction, channel, value);
+}
+
+void SoapyAudio::setGain(const int direction, const size_t channel, const std::string &name, const double value)
+{
+ if (name == "AUDIO")
+ {
+ audioGain = value;
+ SoapySDR_logf(SOAPY_SDR_DEBUG, "Setting Audio Gain: %f", audioGain);
+ }
+}
+
+double SoapyAudio::getGain(const int direction, const size_t channel, const std::string &name) const
+{
+ if ((name.length() >= 2) && (name.substr(0, 2) == "AUDIO"))
+ {
+ return audioGain;
+ }
+
+ return 0;
+}
+
+SoapySDR::Range SoapyAudio::getGainRange(const int direction, const size_t channel, const std::string &name) const
+{
+ return SoapySDR::Range(0, 100);
+}
+
+/*******************************************************************
+ * Frequency API
+ ******************************************************************/
+
+void SoapyAudio::setFrequency(
+ const int direction,
+ const size_t channel,
+ const std::string &name,
+ const double frequency,
+ const SoapySDR::Kwargs &args)
+{
+ if (name == "RF")
+ {
+ centerFrequency = (uint32_t) frequency;
+ resetBuffer = true;
+ SoapySDR_logf(SOAPY_SDR_DEBUG, "Setting center freq: %d", centerFrequency);
+#ifdef USE_HAMLIB
+ if (rigThread && !rigThread->isTerminated()) {
+ if (rigThread->getFrequency() != frequency) {
+ rigThread->setFrequency(frequency);
+ }
+ }
+#endif
+ }
+}
+
+double SoapyAudio::getFrequency(const int direction, const size_t channel, const std::string &name) const
+{
+ if (name == "RF")
+ {
+#ifdef USE_HAMLIB
+ if (rigThread && !rigThread->isTerminated()) {
+ return rigThread->getFrequency();
+ }
+#endif
+ return (double) centerFrequency;
+ }
+
+ return 0;
+}
+
+std::vector<std::string> SoapyAudio::listFrequencies(const int direction, const size_t channel) const
+{
+ std::vector<std::string> names;
+ names.push_back("RF");
+ return names;
+}
+
+SoapySDR::RangeList SoapyAudio::getFrequencyRange(
+ const int direction,
+ const size_t channel,
+ const std::string &name) const
+{
+ SoapySDR::RangeList results;
+ if (name == "RF")
+ {
+ results.push_back(SoapySDR::Range(0, 6000000000));
+ }
+ return results;
+}
+
+SoapySDR::ArgInfoList SoapyAudio::getFrequencyArgsInfo(const int direction, const size_t channel) const
+{
+ SoapySDR::ArgInfoList freqArgs;
+
+ // TODO: frequency arguments
+
+ return freqArgs;
+}
+
+/*******************************************************************
+ * Sample Rate API
+ ******************************************************************/
+
+void SoapyAudio::setSampleRate(const int direction, const size_t channel, const double rate)
+{
+ SoapySDR_logf(SOAPY_SDR_DEBUG, "Setting sample rate: %d", sampleRate);
+
+ if (sampleRate != rate) {
+ sampleRate = rate;
+ resetBuffer = true;
+ sampleRateChanged.store(true);
+ }
+}
+
+double SoapyAudio::getSampleRate(const int direction, const size_t channel) const
+{
+ return sampleRate;
+}
+
+std::vector<double> SoapyAudio::listSampleRates(const int direction, const size_t channel) const
+{
+ std::vector<double> results;
+
+ RtAudio endac;
+ RtAudio::DeviceInfo info = endac.getDeviceInfo(deviceId);
+
+ std::vector<unsigned int>::iterator srate;
+
+ for (srate = info.sampleRates.begin(); srate != info.sampleRates.end(); srate++) {
+ results.push_back(*srate);
+ }
+
+ return results;
+}
+
+void SoapyAudio::setBandwidth(const int direction, const size_t channel, const double bw)
+{
+ SoapySDR::Device::setBandwidth(direction, channel, bw);
+}
+
+double SoapyAudio::getBandwidth(const int direction, const size_t channel) const
+{
+ return SoapySDR::Device::getBandwidth(direction, channel);
+}
+
+std::vector<double> SoapyAudio::listBandwidths(const int direction, const size_t channel) const
+{
+ std::vector<double> results;
+
+ return results;
+}
+
+/*******************************************************************
+ * Settings API
+ ******************************************************************/
+
+SoapySDR::ArgInfoList SoapyAudio::getSettingInfo(void) const
+{
+ SoapySDR::ArgInfoList setArgs;
+
+ // Sample Offset
+ SoapySDR::ArgInfo sampleOffsetArg;
+ sampleOffsetArg.key = "sample_offset";
+ sampleOffsetArg.value = "0";
+ sampleOffsetArg.name = "Stereo Sample Offset";
+ sampleOffsetArg.description = "Offset stereo samples for off-by-one audio inputs.";
+ sampleOffsetArg.type = SoapySDR::ArgInfo::STRING;
+
+ std::vector<std::string> sampleOffsetOpts;
+ std::vector<std::string> sampleOffsetOptNames;
+
+ sampleOffsetOpts.push_back("-2");
+ sampleOffsetOptNames.push_back("-2 Samples");
+ sampleOffsetOpts.push_back("-1");
+ sampleOffsetOptNames.push_back("-1 Samples");
+ sampleOffsetOpts.push_back("0");
+ sampleOffsetOptNames.push_back("0 Samples");
+ sampleOffsetOpts.push_back("1");
+ sampleOffsetOptNames.push_back("1 Samples");
+ sampleOffsetOpts.push_back("2");
+ sampleOffsetOptNames.push_back("2 Samples");
+
+ sampleOffsetArg.options = sampleOffsetOpts;
+ sampleOffsetArg.optionNames = sampleOffsetOptNames;
+
+ setArgs.push_back(sampleOffsetArg);
+
+#ifdef USE_HAMLIB
+ // Rig Control
+ SoapySDR::ArgInfo rigArg;
+ rigArg.key = "rig";
+ rigArg.value = "";
+ rigArg.name = "Rig Control";
+ rigArg.description = "Select hamlib rig control type.";
+ rigArg.type = SoapySDR::ArgInfo::STRING;
+
+ std::vector<std::string> rigOpts;
+ std::vector<std::string> rigOptNames;
+
+ rigOpts.push_back("");
+ rigOptNames.push_back("None");
+
+ for (std::vector<const struct rig_caps *>::const_iterator i = rigCaps.begin(); i != rigCaps.end(); i++) {
+ const struct rig_caps *rc = (*i);
+
+ rigOpts.push_back(std::to_string(rc->rig_model));
+ rigOptNames.push_back(std::string(rc->mfg_name) + " " + std::string(rc->model_name));
+ }
+
+ rigArg.options = rigOpts;
+ rigArg.optionNames = rigOptNames;
+
+ setArgs.push_back(rigArg);
+
+ // Rig Control
+ SoapySDR::ArgInfo rigRateArg;
+ rigRateArg.key = "rig_rate";
+ rigRateArg.value = "57600";
+ rigRateArg.name = "Rig Serial Rate";
+ rigRateArg.description = "Select hamlib rig serial control rate.";
+ rigRateArg.type = SoapySDR::ArgInfo::STRING;
+
+ std::vector<std::string> rigRateOpts;
+ std::vector<std::string> rigRateOptNames;
+
+ rigRateOpts.push_back("1200");
+ rigRateOptNames.push_back("1200 baud");
+ rigRateOpts.push_back("2400");
+ rigRateOptNames.push_back("2400 baud");
+ rigRateOpts.push_back("4800");
+ rigRateOptNames.push_back("4800 baud");
+ rigRateOpts.push_back("9600");
+ rigRateOptNames.push_back("9600 baud");
+ rigRateOpts.push_back("19200");
+ rigRateOptNames.push_back("19200 baud");
+ rigRateOpts.push_back("38400");
+ rigRateOptNames.push_back("38400 baud");
+ rigRateOpts.push_back("57600");
+ rigRateOptNames.push_back("57600 baud");
+ rigRateOpts.push_back("115200");
+ rigRateOptNames.push_back("115200 baud");
+ rigRateOpts.push_back("128000");
+ rigRateOptNames.push_back("128000 baud");
+ rigRateOpts.push_back("256000");
+ rigRateOptNames.push_back("256000 baud");
+
+ rigRateArg.options = rigRateOpts;
+ rigRateArg.optionNames = rigRateOptNames;
+
+ setArgs.push_back(rigRateArg);
+
+ SoapySDR::ArgInfo rigFileArg;
+ rigFileArg.key = "rig_port";
+ rigFileArg.value = "/dev/ttyUSB0";
+ rigFileArg.name = "Rig Serial Port";
+ rigFileArg.description = "hamlib rig Serial Port dev / COMx / IP-Address";
+ rigFileArg.type = SoapySDR::ArgInfo::STRING;
+
+ setArgs.push_back(rigFileArg);
+
+#endif
+
+ return setArgs;
+}
+
+void SoapyAudio::writeSetting(const std::string &key, const std::string &value)
+{
+ if (key == "sample_offset") {
+ try {
+ int sOffset = std::stoi(value);
+
+ if (sOffset >= -2 && sOffset <= 2) {
+ sampleOffset = sOffset;
+ }
+ } catch (std::invalid_argument e) { }
+ }
+
+
+#ifdef USE_HAMLIB
+ bool rigReset = false;
+ if (key == "rig")
+ {
+ try {
+ rig_model_t newModel = std::stoi(value);
+ if (newModel != rigModel) {
+ rigReset = true;
+ rigModel = newModel;
+ }
+ } catch (const std::invalid_argument &) {
+ rigModel = 0;
+ }
+ }
+
+ if (key == "rig_rate")
+ {
+ try {
+ int newSerialRate = std::stoi(value);
+ if (newSerialRate != rigSerialRate) {
+ rigSerialRate = newSerialRate;
+ rigReset = true;
+ }
+ } catch (const std::invalid_argument &) {
+ rigSerialRate = 57600;
+ }
+ }
+
+ if (key == "rig_port")
+ {
+ if (rigFile != value) {
+ rigFile = value;
+ rigReset = true;
+ }
+ }
+
+ if (rigReset) {
+ if (rigThread && !rigThread->isTerminated()) {
+ rigThread->terminate();
+ }
+ checkRigThread();
+ }
+#endif
+}
+
+std::string SoapyAudio::readSetting(const std::string &key) const
+{
+ if (key == "sample_offset") {
+ return std::to_string(sampleOffset);
+ }
+
+#ifdef USE_HAMLIB
+ if (key == "rig")
+ {
+ return std::to_string(rigModel);
+ }
+ if (key == "rig_rate")
+ {
+ return std::to_string(rigSerialRate);
+ }
+ if (key == "rig_port")
+ {
+ return rigFile;
+ }
+#endif
+ // SoapySDR_logf(SOAPY_SDR_WARNING, "Unknown setting '%s'", key.c_str());
+ return "";
+}
+
+
+chanSetup SoapyAudio::chanSetupStrToEnum(std::string chanOpt) {
+ if (chanOpt == "mono_l") {
+ return FORMAT_MONO_L;
+ } else if (chanOpt == "mono_r") {
+ return FORMAT_MONO_R;
+ } else if (chanOpt == "stereo_iq") {
+ return FORMAT_STEREO_IQ;
+ } else if (chanOpt == "stereo_qi") {
+ return FORMAT_STEREO_QI;
+ } else {
+ return FORMAT_MONO_L;
+ }
+}
+
+#ifdef USE_HAMLIB
+void SoapyAudio::checkRigThread() {
+ if (!rigModel || !rigSerialRate || rigFile == "") {
+ return;
+ }
+ if (!rigThread) {
+ rigThread = new RigThread();
+ }
+ if (rigThread->isTerminated()) {
+ if (t_Rig && t_Rig->joinable()) {
+ t_Rig->join();
+ delete t_Rig;
+ }
+ rigThread->setup(rigModel, rigFile, rigSerialRate);
+ t_Rig = new std::thread(&RigThread::threadMain, rigThread);
+ }
+}
+
+#endif
\ No newline at end of file
diff --git a/SoapyAudio.hpp b/SoapyAudio.hpp
new file mode 100644
index 0000000..6eefb76
--- /dev/null
+++ b/SoapyAudio.hpp
@@ -0,0 +1,275 @@
+/*
+ * The MIT License (MIT)
+ *
+ * Copyright (c) 2015 Charles J. Cliffe
+
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+#pragma once
+
+#include <SoapySDR/Device.hpp>
+#include <SoapySDR/Logger.h>
+#include <SoapySDR/Types.h>
+#include <RtAudio.h>
+#include <stdexcept>
+#include <thread>
+#include <mutex>
+#include <atomic>
+#include <condition_variable>
+#include <string>
+#include <cstring>
+#include <algorithm>
+
+#ifdef USE_HAMLIB
+#include "RigThread.h"
+#endif
+
+typedef enum audioStreamFormat
+{
+ AUDIO_FORMAT_FLOAT32, AUDIO_FORMAT_INT16, AUDIO_FORMAT_INT8
+} audioStreamFormat;
+
+typedef enum chanSetup
+{
+ FORMAT_MONO_L, FORMAT_MONO_R, FORMAT_STEREO_IQ, FORMAT_STEREO_QI
+} chanSetup;
+
+#define DEFAULT_BUFFER_LENGTH 2048
+#define DEFAULT_NUM_BUFFERS 6
+
+class SoapyAudio: public SoapySDR::Device
+{
+public:
+ SoapyAudio(const SoapySDR::Kwargs &args);
+
+ ~SoapyAudio(void);
+
+ /*******************************************************************
+ * Identification API
+ ******************************************************************/
+
+ std::string getDriverKey(void) const;
+
+ std::string getHardwareKey(void) const;
+
+ SoapySDR::Kwargs getHardwareInfo(void) const;
+
+ /*******************************************************************
+ * Channels API
+ ******************************************************************/
+
+ size_t getNumChannels(const int) const;
+
+ /*******************************************************************
+ * Stream API
+ ******************************************************************/
+
+ std::vector<std::string> getStreamFormats(const int direction, const size_t channel) const;
+
+ std::string getNativeStreamFormat(const int direction, const size_t channel, double &fullScale) const;
+
+ SoapySDR::ArgInfoList getStreamArgsInfo(const int direction, const size_t channel) const;
+
+ SoapySDR::Stream *setupStream(const int direction, const std::string &format, const std::vector<size_t> &channels =
+ std::vector<size_t>(), const SoapySDR::Kwargs &args = SoapySDR::Kwargs());
+
+ void closeStream(SoapySDR::Stream *stream);
+
+ size_t getStreamMTU(SoapySDR::Stream *stream) const;
+
+ int activateStream(
+ SoapySDR::Stream *stream,
+ const int flags = 0,
+ const long long timeNs = 0,
+ const size_t numElems = 0);
+
+ int deactivateStream(SoapySDR::Stream *stream, const int flags = 0, const long long timeNs = 0);
+
+ int readStream(
+ SoapySDR::Stream *stream,
+ void * const *buffs,
+ const size_t numElems,
+ int &flags,
+ long long &timeNs,
+ const long timeoutUs = 100000);
+
+ /*******************************************************************
+ * Direct buffer access API
+ ******************************************************************/
+
+ size_t getNumDirectAccessBuffers(SoapySDR::Stream *stream);
+
+ int getDirectAccessBufferAddrs(SoapySDR::Stream *stream, const size_t handle, void **buffs);
+
+ int acquireReadBuffer(
+ SoapySDR::Stream *stream,
+ size_t &handle,
+ const void **buffs,
+ int &flags,
+ long long &timeNs,
+ const long timeoutUs = 100000);
+
+ void releaseReadBuffer(
+ SoapySDR::Stream *stream,
+ const size_t handle);
+
+ /*******************************************************************
+ * Antenna API
+ ******************************************************************/
+
+ std::vector<std::string> listAntennas(const int direction, const size_t channel) const;
+
+ void setAntenna(const int direction, const size_t channel, const std::string &name);
+
+ std::string getAntenna(const int direction, const size_t channel) const;
+
+ /*******************************************************************
+ * Frontend corrections API
+ ******************************************************************/
+
+ bool hasDCOffsetMode(const int direction, const size_t channel) const;
+
+ /*******************************************************************
+ * Gain API
+ ******************************************************************/
+
+ std::vector<std::string> listGains(const int direction, const size_t channel) const;
+
+ bool hasGainMode(const int direction, const size_t channel) const;
+
+ void setGainMode(const int direction, const size_t channel, const bool automatic);
+
+ bool getGainMode(const int direction, const size_t channel) const;
+
+ void setGain(const int direction, const size_t channel, const double value);
+
+ void setGain(const int direction, const size_t channel, const std::string &name, const double value);
+
+ double getGain(const int direction, const size_t channel, const std::string &name) const;
+
+ SoapySDR::Range getGainRange(const int direction, const size_t channel, const std::string &name) const;
+
+ /*******************************************************************
+ * Frequency API
+ ******************************************************************/
+
+ void setFrequency(
+ const int direction,
+ const size_t channel,
+ const std::string &name,
+ const double frequency,
+ const SoapySDR::Kwargs &args = SoapySDR::Kwargs());
+
+ double getFrequency(const int direction, const size_t channel, const std::string &name) const;
+
+ std::vector<std::string> listFrequencies(const int direction, const size_t channel) const;
+
+ SoapySDR::RangeList getFrequencyRange(const int direction, const size_t channel, const std::string &name) const;
+
+ SoapySDR::ArgInfoList getFrequencyArgsInfo(const int direction, const size_t channel) const;
+
+ /*******************************************************************
+ * Sample Rate API
+ ******************************************************************/
+
+ void setSampleRate(const int direction, const size_t channel, const double rate);
+
+ double getSampleRate(const int direction, const size_t channel) const;
+
+ std::vector<double> listSampleRates(const int direction, const size_t channel) const;
+
+ void setBandwidth(const int direction, const size_t channel, const double bw);
+
+ double getBandwidth(const int direction, const size_t channel) const;
+
+ std::vector<double> listBandwidths(const int direction, const size_t channel) const;
+
+ /*******************************************************************
+ * Utility
+ ******************************************************************/
+
+ chanSetup chanSetupStrToEnum(std::string chanOpt);
+
+ /*******************************************************************
+ * Settings API
+ ******************************************************************/
+
+ SoapySDR::ArgInfoList getSettingInfo(void) const;
+
+ void writeSetting(const std::string &key, const std::string &value);
+
+ std::string readSetting(const std::string &key) const;
+
+private:
+
+ //device handle
+ int deviceId;
+ RtAudio dac;
+ RtAudio::DeviceInfo devInfo;
+ RtAudio::StreamOptions opts;
+ RtAudio::StreamParameters inputParameters;
+ RtAudio::StreamParameters outputParameters;
+
+ //cached settings
+ audioStreamFormat asFormat;
+ chanSetup cSetup;
+ uint32_t sampleRate, centerFrequency;
+ unsigned int bufferLength;
+ size_t numBuffers;
+ bool agcMode, streamActive;
+ std::atomic_bool sampleRateChanged;
+ double audioGain;
+ int elementsPerSample;
+ int sampleOffset;
+ float sampleOffsetBuffer[2];
+
+public:
+ //async api usage
+ int rx_callback(void *inputBuffer, unsigned int nBufferFrames, double streamTime, RtAudioStreamStatus status);
+
+ std::mutex _buf_mutex;
+ std::condition_variable _buf_cond;
+
+ std::vector<std::vector<float> > _buffs;
+ size_t _buf_head;
+ size_t _buf_tail;
+ size_t _buf_count;
+ float *_currentBuff;
+ bool _overflowEvent;
+ size_t _currentHandle;
+ size_t bufferedElems;
+ bool resetBuffer;
+
+
+#ifdef USE_HAMLIB
+public:
+ static int add_hamlib_rig(const struct rig_caps *rc, void* f);
+ static std::vector<const struct rig_caps *> rigCaps;
+
+ void checkRigThread();
+
+private:
+ RigThread *rigThread;
+ std::thread *t_Rig;
+
+ rig_model_t rigModel;
+ std::string rigFile;
+ int rigSerialRate;
+#endif
+};
diff --git a/Streaming.cpp b/Streaming.cpp
new file mode 100644
index 0000000..f6a9027
--- /dev/null
+++ b/Streaming.cpp
@@ -0,0 +1,678 @@
+/*
+ * The MIT License (MIT)
+ *
+ * Copyright (c) 2015 Charles J. Cliffe
+
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+#include "SoapyAudio.hpp"
+#include <SoapySDR/Logger.hpp>
+#include <algorithm> //min
+#include <climits> //SHRT_MAX
+#include <cstring> // memcpy
+
+
+std::vector<std::string> SoapyAudio::getStreamFormats(const int direction, const size_t channel) const {
+ std::vector<std::string> formats;
+
+ formats.push_back("CS8");
+ formats.push_back("CS16");
+ formats.push_back("CF32");
+
+ return formats;
+}
+
+std::string SoapyAudio::getNativeStreamFormat(const int direction, const size_t channel, double &fullScale) const {
+ fullScale = 65536;
+ return "CS16";
+}
+
+SoapySDR::ArgInfoList SoapyAudio::getStreamArgsInfo(const int direction, const size_t channel) const {
+ SoapySDR::ArgInfoList streamArgs;
+
+ SoapySDR::ArgInfo chanArg;
+ chanArg.key = "chan";
+ chanArg.value = "mono_l";
+ chanArg.name = "Channel Setup";
+ chanArg.description = "Input channel configuration.";
+ chanArg.type = SoapySDR::ArgInfo::STRING;
+
+ std::vector<std::string> chanOpts;
+ std::vector<std::string> chanOptNames;
+
+ chanOpts.push_back("mono_l");
+ chanOptNames.push_back("Mono Left");
+ chanOpts.push_back("mono_r");
+ chanOptNames.push_back("Mono Right");
+ chanOpts.push_back("stereo_iq");
+ chanOptNames.push_back("Complex L/R = I/Q");
+ chanOpts.push_back("stereo_qi");
+ chanOptNames.push_back("Complex L/R = Q/I");
+
+ chanArg.options = chanOpts;
+ chanArg.optionNames = chanOptNames;
+
+ streamArgs.push_back(chanArg);
+
+ return streamArgs;
+}
+
+/*******************************************************************
+ * Async thread work
+ ******************************************************************/
+
+
+static int _rx_callback(void *outputBuffer, void *inputBuffer, unsigned int nBufferFrames, double streamTime, RtAudioStreamStatus status,
+ void *ctx)
+{
+ //printf("_rx_callback\n");
+ SoapyAudio *self = (SoapyAudio *)ctx;
+ return self->rx_callback(inputBuffer, nBufferFrames, streamTime, status);
+}
+
+int SoapyAudio::rx_callback(void *inputBuffer, unsigned int nBufferFrames, double streamTime, RtAudioStreamStatus status)
+{
+ std::unique_lock<std::mutex> lock(_buf_mutex);
+
+ if (sampleRateChanged.load()) {
+ return 1;
+ }
+
+ //printf("_rx_callback %d _buf_head=%d, numBuffers=%d\n", len, _buf_head, _buf_tail);
+
+ //overflow condition: the caller is not reading fast enough
+ if (_buf_count == numBuffers)
+ {
+ _overflowEvent = true;
+ return 0;
+ }
+
+ //copy into the buffer queue
+ auto &buff = _buffs[_buf_tail];
+ buff.resize(nBufferFrames * elementsPerSample);
+ std::memcpy(buff.data(), inputBuffer, nBufferFrames * elementsPerSample * sizeof(float));
+
+ //increment the tail pointer
+ _buf_tail = (_buf_tail + 1) % numBuffers;
+ _buf_count++;
+
+ //notify readStream()
+ _buf_cond.notify_one();
+
+ return 0;
+}
+
+/*******************************************************************
+ * Stream API
+ ******************************************************************/
+
+SoapySDR::Stream *SoapyAudio::setupStream(
+ const int direction,
+ const std::string &format,
+ const std::vector<size_t> &channels,
+ const SoapySDR::Kwargs &args)
+{
+ //check the channel configuration
+ if (channels.size() > 1 or (channels.size() > 0 and channels.at(0) != 0))
+ {
+ throw std::runtime_error("setupStream invalid channel selection");
+ }
+
+ //check the format
+ if (format == "CF32")
+ {
+ SoapySDR_log(SOAPY_SDR_INFO, "Using format CF32.");
+ asFormat = AUDIO_FORMAT_FLOAT32;
+ }
+ else if (format == "CS16")
+ {
+ SoapySDR_log(SOAPY_SDR_INFO, "Using format CS16.");
+ asFormat = AUDIO_FORMAT_INT16;
+ }
+ else if (format == "CS8") {
+ SoapySDR_log(SOAPY_SDR_INFO, "Using format CS8.");
+ asFormat = AUDIO_FORMAT_INT8;
+ }
+ else
+ {
+ throw std::runtime_error(
+ "setupStream invalid format '" + format
+ + "' -- Only CS8, CS16 and CF32 are supported by SoapyAudio module.");
+ }
+
+ if (args.count("chan") != 0)
+ {
+ std::string chanOpt = args.at("chan");
+ cSetup = chanSetupStrToEnum(chanOpt);
+ } else {
+ cSetup = FORMAT_MONO_L;
+ }
+
+ inputParameters.deviceId = deviceId;
+
+ switch (cSetup) {
+ case FORMAT_MONO_L:
+ inputParameters.nChannels = 1;
+ inputParameters.firstChannel = 0;
+ bufferLength = DEFAULT_BUFFER_LENGTH;
+ elementsPerSample = 1;
+ break;
+ case FORMAT_MONO_R:
+ inputParameters.nChannels = 1;
+ inputParameters.firstChannel = 1;
+ bufferLength = DEFAULT_BUFFER_LENGTH;
+ elementsPerSample = 1;
+ break;
+ case FORMAT_STEREO_IQ:
+ inputParameters.nChannels = 2;
+ inputParameters.firstChannel = 0;
+ bufferLength = DEFAULT_BUFFER_LENGTH*2;
+ elementsPerSample = 2;
+ break;
+ case FORMAT_STEREO_QI:
+ inputParameters.nChannels = 2;
+ inputParameters.firstChannel = 0;
+ bufferLength = DEFAULT_BUFFER_LENGTH*2;
+ elementsPerSample = 2;
+ break;
+ }
+
+ //clear async fifo counts
+ _buf_tail = 0;
+ _buf_count = 0;
+ _buf_head = 0;
+
+ //allocate buffers
+ _buffs.resize(numBuffers);
+ for (auto &buff : _buffs) buff.reserve(bufferLength);
+ for (auto &buff : _buffs) buff.resize(bufferLength);
+
+ return (SoapySDR::Stream *) this;
+}
+
+void SoapyAudio::closeStream(SoapySDR::Stream *stream)
+{
+ _buffs.clear();
+}
+
+size_t SoapyAudio::getStreamMTU(SoapySDR::Stream *stream) const
+{
+ return bufferLength / elementsPerSample;
+}
+
+int SoapyAudio::activateStream(
+ SoapySDR::Stream *stream,
+ const int flags,
+ const long long timeNs,
+ const size_t numElems)
+{
+ if (flags != 0) return SOAPY_SDR_NOT_SUPPORTED;
+ resetBuffer = true;
+ bufferedElems = 0;
+
+ try {
+#ifndef _MSC_VER
+ opts.priority = sched_get_priority_max(SCHED_FIFO);
+#endif
+ // opts.flags = RTAUDIO_MINIMIZE_LATENCY;
+ opts.flags = RTAUDIO_SCHEDULE_REALTIME;
+
+ sampleRateChanged.store(false);
+ dac.openStream(NULL, &inputParameters, RTAUDIO_FLOAT32, sampleRate, &bufferLength, &_rx_callback, (void *) this, &opts);
+ dac.startStream();
+
+ streamActive = true;
+ } catch (RtAudioError& e) {
+ throw std::runtime_error("RtAudio init error '" + e.getMessage());
+ }
+
+ return 0;
+}
+
+int SoapyAudio::deactivateStream(SoapySDR::Stream *stream, const int flags, const long long timeNs)
+{
+ if (flags != 0) return SOAPY_SDR_NOT_SUPPORTED;
+
+ if (dac.isStreamRunning()) {
+ dac.stopStream();
+ }
+ if (dac.isStreamOpen()) {
+ dac.closeStream();
+ }
+
+ streamActive = false;
+
+ return 0;
+}
+
+int SoapyAudio::readStream(
+ SoapySDR::Stream *stream,
+ void * const *buffs,
+ const size_t numElems,
+ int &flags,
+ long long &timeNs,
+ const long timeoutUs)
+{
+ if (!dac.isStreamRunning()) {
+ return 0;
+ }
+
+ if (sampleRateChanged.load()) {
+ if (dac.isStreamRunning()) {
+ dac.stopStream();
+ }
+ if (dac.isStreamOpen()) {
+ dac.closeStream();
+ }
+ dac.openStream(NULL, &inputParameters, RTAUDIO_FLOAT32, sampleRate, &bufferLength, &_rx_callback, (void *) this, &opts);
+ dac.startStream();
+ sampleRateChanged.store(false);
+ }
+
+ //this is the user's buffer for channel 0
+ void *buff0 = buffs[0];
+
+ //are elements left in the buffer? if not, do a new read.
+ if (bufferedElems == 0 || (sampleOffset && (bufferedElems < abs(sampleOffset))))
+ {
+ int ret = this->acquireReadBuffer(stream, _currentHandle, (const void **)&_currentBuff, flags, timeNs, timeoutUs);
+ if (ret < 0) return ret;
+ bufferedElems = ret;
+ }
+
+ size_t returnedElems = std::min(bufferedElems, numElems);
+
+ if (sampleOffset && (bufferedElems < abs(sampleOffset))) {
+ return 0;
+ }
+
+ //convert into user's buff0
+ if (sampleOffset) {
+ if (asFormat == AUDIO_FORMAT_FLOAT32)
+ {
+ float *ftarget = (float *) buff0;
+ std::complex<float> tmp;
+ if (cSetup == FORMAT_MONO_L || cSetup == FORMAT_MONO_R) {
+ for (size_t i = 0; i < returnedElems; i++)
+ {
+ ftarget[i * 2] = _currentBuff[i];
+ ftarget[i * 2 + 1] = 0;
+ }
+ }
+ else if (cSetup == FORMAT_STEREO_IQ) {
+ if (sampleOffset > 0) {
+ size_t iStart = abs(sampleOffset);
+ for (size_t i = 0; i < iStart; i++) {
+ ftarget[i * 2] = sampleOffsetBuffer[i];
+ ftarget[i * 2 + 1] = _currentBuff[i * 2 + 1];
+ }
+ for (size_t i = iStart; i < returnedElems; i++) {
+ ftarget[i * 2] = _currentBuff[(i + iStart) * 2];
+ ftarget[i * 2 + 1] = _currentBuff[i * 2 + 1];
+ }
+ for (int i = 0; i < iStart; i++) {
+ sampleOffsetBuffer[i] = _currentBuff[(returnedElems-iStart+i) * 2];
+ }
+ } else {
+ size_t iStart = abs(sampleOffset);
+ for (size_t i = 0; i < iStart; i++) {
+ ftarget[i * 2] = _currentBuff[i * 2];
+ ftarget[i * 2 + 1] = sampleOffsetBuffer[i];
+ }
+ for (size_t i = iStart; i < returnedElems; i++) {
+ ftarget[i * 2] = _currentBuff[i * 2];
+ ftarget[i * 2 + 1] = _currentBuff[(i + iStart) * 2 + 1];
+ }
+ for (int i = 0; i < iStart; i++) {
+ sampleOffsetBuffer[i] = _currentBuff[(returnedElems-iStart+i) * 2 + 1];
+ }
+ }
+ }
+ else if (cSetup == FORMAT_STEREO_QI) {
+ if (sampleOffset > 0) {
+ size_t iStart = abs(sampleOffset);
+ for (size_t i = 0; i < iStart; i++) {
+ ftarget[i * 2 + 1] = sampleOffsetBuffer[i];
+ ftarget[i * 2] = _currentBuff[i * 2 + 1];
+ }
+ for (size_t i = iStart; i < returnedElems; i++) {
+ ftarget[i * 2 + 1] = _currentBuff[(i + iStart) * 2];
+ ftarget[i * 2] = _currentBuff[i * 2 + 1];
+ }
+ for (int i = 0; i < iStart; i++) {
+ sampleOffsetBuffer[i] = _currentBuff[(returnedElems-iStart+i) * 2];
+ }
+ } else {
+ size_t iStart = abs(sampleOffset);
+ for (size_t i = 0; i < iStart; i++) {
+ ftarget[i * 2 + 1] = _currentBuff[i * 2];
+ ftarget[i * 2] = sampleOffsetBuffer[i];
+ }
+ for (size_t i = iStart; i < returnedElems; i++) {
+ ftarget[i * 2 + 1] = _currentBuff[i * 2];
+ ftarget[i * 2] = _currentBuff[(i + iStart) * 2 + 1];
+ }
+ for (int i = 0; i < iStart; i++) {
+ sampleOffsetBuffer[i] = _currentBuff[(returnedElems-iStart+i) * 2 + 1];
+ }
+ }
+ }
+ }
+ else if (asFormat == AUDIO_FORMAT_INT16)
+ {
+ int16_t *itarget = (int16_t *) buff0;
+ std::complex<int16_t> tmp;
+ if (cSetup == FORMAT_MONO_L || cSetup == FORMAT_MONO_R) {
+ for (size_t i = 0; i < returnedElems; i++)
+ {
+ itarget[i * 2] = int16_t(_currentBuff[i] * 32767.0);
+ itarget[i * 2 + 1] = 0;
+ }
+ }
+ else if (cSetup == FORMAT_STEREO_IQ) {
+ if (sampleOffset > 0) {
+ size_t iStart = abs(sampleOffset);
+ for (size_t i = 0; i < iStart; i++) {
+ itarget[i * 2] = int16_t(sampleOffsetBuffer[i] * 32767.0);
+ itarget[i * 2 + 1] = int16_t(_currentBuff[i * 2 + 1] * 32767.0);
+ }
+ for (size_t i = iStart; i < returnedElems; i++) {
+ itarget[i * 2] = int16_t(_currentBuff[(i + iStart) * 2] * 32767.0);
+ itarget[i * 2 + 1] = int16_t(_currentBuff[i * 2 + 1] * 32767.0);
+ }
+ for (int i = 0; i < iStart; i++) {
+ sampleOffsetBuffer[i] = _currentBuff[(returnedElems-iStart+i) * 2];
+ }
+ } else {
+ size_t iStart = abs(sampleOffset);
+ for (size_t i = 0; i < iStart; i++) {
+ itarget[i * 2] = int16_t(_currentBuff[i * 2] * 32767.0);
+ itarget[i * 2 + 1] = int16_t(sampleOffsetBuffer[i] * 32767.0);
+ }
+ for (size_t i = iStart; i < returnedElems; i++) {
+ itarget[i * 2] = int16_t(_currentBuff[i * 2] * 32767.0);
+ itarget[i * 2 + 1] = int16_t(_currentBuff[(i + iStart) * 2 + 1] * 32767.0);
+ }
+ for (int i = 0; i < iStart; i++) {
+ sampleOffsetBuffer[i] = _currentBuff[(returnedElems-iStart+i) * 2 + 1];
+ }
+ }
+ }
+ if (cSetup == FORMAT_STEREO_QI) {
+ if (sampleOffset > 0) {
+ size_t iStart = abs(sampleOffset);
+ for (size_t i = 0; i < iStart; i++) {
+ itarget[i * 2 + 1] = int16_t(sampleOffsetBuffer[i] * 32767.0);
+ itarget[i * 2] = int16_t(_currentBuff[i * 2 + 1] * 32767.0);
+ }
+ for (size_t i = iStart; i < returnedElems; i++) {
+ itarget[i * 2 + 1] = int16_t(_currentBuff[(i + iStart) * 2] * 32767.0);
+ itarget[i * 2] = int16_t(_currentBuff[i * 2 + 1] * 32767.0);
+ }
+ for (int i = 0; i < iStart; i++) {
+ sampleOffsetBuffer[i] = _currentBuff[(returnedElems-iStart+i) * 2];
+ }
+ } else {
+ size_t iStart = abs(sampleOffset);
+ for (size_t i = 0; i < iStart; i++) {
+ itarget[i * 2 + 1] = int16_t(_currentBuff[i * 2] * 32767.0);
+ itarget[i * 2] = int16_t(sampleOffsetBuffer[i] * 32767.0);
+ }
+ for (size_t i = iStart; i < returnedElems; i++) {
+ itarget[i * 2 + 1] = int16_t(_currentBuff[i * 2] * 32767.0);
+ itarget[i * 2] = int16_t(_currentBuff[(i + iStart) * 2 + 1] * 32767.0);
+ }
+ for (int i = 0; i < iStart; i++) {
+ sampleOffsetBuffer[i] = _currentBuff[(returnedElems-iStart+i) * 2 + 1];
+ }
+ }
+ }
+ }
+ else if (asFormat == AUDIO_FORMAT_INT8)
+ {
+ int8_t *itarget = (int8_t *) buff0;
+ if (cSetup == FORMAT_MONO_L || cSetup == FORMAT_MONO_R) {
+ for (size_t i = 0; i < returnedElems; i++)
+ {
+ itarget[i * 2] = int8_t(_currentBuff[i] * 127.0);
+ itarget[i * 2 + 1] = 0;
+ }
+ }
+ else if (cSetup == FORMAT_STEREO_IQ) {
+ if (sampleOffset > 0) {
+ size_t iStart = abs(sampleOffset);
+ for (size_t i = 0; i < iStart; i++) {
+ itarget[i * 2] = int16_t(sampleOffsetBuffer[i] * 127.0);
+ itarget[i * 2 + 1] = int16_t(_currentBuff[i * 2 + 1] * 127.0);
+ }
+ for (size_t i = iStart; i < returnedElems; i++) {
+ itarget[i * 2] = int16_t(_currentBuff[(i + iStart) * 2] * 127.0);
+ itarget[i * 2 + 1] = int16_t(_currentBuff[i * 2 + 1] * 127.0);
+ }
+ for (int i = 0; i < iStart; i++) {
+ sampleOffsetBuffer[i] = _currentBuff[(returnedElems-iStart+i) * 2];
+ }
+ } else {
+ size_t iStart = abs(sampleOffset);
+ for (size_t i = 0; i < iStart; i++) {
+ itarget[i * 2] = int16_t(_currentBuff[i * 2] * 127.0);
+ itarget[i * 2 + 1] = int16_t(sampleOffsetBuffer[i] * 127.0);
+ }
+ for (size_t i = iStart; i < returnedElems; i++) {
+ itarget[i * 2] = int16_t(_currentBuff[i * 2] * 127.0);
+ itarget[i * 2 + 1] = int16_t(_currentBuff[(i + iStart) * 2 + 1] * 127.0);
+ }
+ for (int i = 0; i < iStart; i++) {
+ sampleOffsetBuffer[i] = _currentBuff[(returnedElems-iStart+i) * 2 + 1];
+ }
+ }
+ }
+ if (cSetup == FORMAT_STEREO_QI) {
+ if (sampleOffset > 0) {
+ size_t iStart = abs(sampleOffset);
+ for (size_t i = 0; i < iStart; i++) {
+ itarget[i * 2 + 1] = int16_t(sampleOffsetBuffer[i] * 127.0);
+ itarget[i * 2] = int16_t(_currentBuff[i * 2 + 1] * 127.0);
+ }
+ for (size_t i = iStart; i < returnedElems; i++) {
+ itarget[i * 2 + 1] = int16_t(_currentBuff[(i + iStart) * 2] * 127.0);
+ itarget[i * 2] = int16_t(_currentBuff[i * 2 + 1] * 127.0);
+ }
+ for (int i = 0; i < iStart; i++) {
+ sampleOffsetBuffer[i] = _currentBuff[(returnedElems-iStart+i) * 2];
+ }
+ } else {
+ size_t iStart = abs(sampleOffset);
+ for (size_t i = 0; i < iStart; i++) {
+ itarget[i * 2 + 1] = int16_t(_currentBuff[i * 2] * 127.0);
+ itarget[i * 2] = int16_t(sampleOffsetBuffer[i] * 127.0);
+ }
+ for (size_t i = iStart; i < returnedElems; i++) {
+ itarget[i * 2 + 1] = int16_t(_currentBuff[i * 2] * 127.0);
+ itarget[i * 2] = int16_t(_currentBuff[(i + iStart) * 2 + 1] * 127.0);
+ }
+ for (int i = 0; i < iStart; i++) {
+ sampleOffsetBuffer[i] = _currentBuff[(returnedElems-iStart+i) * 2 + 1];
+ }
+ }
+ }
+ }
+ } else {
+ if (asFormat == AUDIO_FORMAT_FLOAT32)
+ {
+ float *ftarget = (float *) buff0;
+ std::complex<float> tmp;
+ if (cSetup == FORMAT_MONO_L || cSetup == FORMAT_MONO_R) {
+ for (size_t i = 0; i < returnedElems; i++)
+ {
+ ftarget[i * 2] = _currentBuff[i];
+ ftarget[i * 2 + 1] = 0;
+ }
+ }
+ else if (cSetup == FORMAT_STEREO_IQ) {
+ for (size_t i = 0; i < returnedElems; i++)
+ {
+ ftarget[i * 2] = _currentBuff[i * 2];
+ ftarget[i * 2 + 1] = _currentBuff[i * 2 + 1];
+ }
+ }
+ else if (cSetup == FORMAT_STEREO_QI) {
+ for (size_t i = 0; i < returnedElems; i++)
+ {
+ ftarget[i * 2] = _currentBuff[i * 2 + 1];
+ ftarget[i * 2 + 1] = _currentBuff[i * 2];
+ }
+ }
+ }
+ else if (asFormat == AUDIO_FORMAT_INT16)
+ {
+ int16_t *itarget = (int16_t *) buff0;
+ std::complex<int16_t> tmp;
+ if (cSetup == FORMAT_MONO_L || cSetup == FORMAT_MONO_R) {
+ for (size_t i = 0; i < returnedElems; i++)
+ {
+ itarget[i * 2] = int16_t(_currentBuff[i] * 32767.0);
+ itarget[i * 2 + 1] = 0;
+ }
+ }
+ else if (cSetup == FORMAT_STEREO_IQ) {
+ for (size_t i = 0; i < returnedElems; i++)
+ {
+ itarget[i * 2] = int16_t(_currentBuff[i * 2] * 32767.0);
+ itarget[i * 2 + 1] = int16_t(_currentBuff[i * 2 + 1] * 32767.0);
+ }
+ }
+ else if (cSetup == FORMAT_STEREO_QI) {
+ for (size_t i = 0; i < returnedElems; i++)
+ {
+ itarget[i * 2] = int16_t(_currentBuff[i * 2 + 1] * 32767.0);
+ itarget[i * 2 + 1] = int16_t(_currentBuff[i * 2] * 32767.0);
+ }
+ }
+ }
+ else if (asFormat == AUDIO_FORMAT_INT8)
+ {
+ int8_t *itarget = (int8_t *) buff0;
+ if (cSetup == FORMAT_MONO_L || cSetup == FORMAT_MONO_R) {
+ for (size_t i = 0; i < returnedElems; i++)
+ {
+ itarget[i * 2] = int8_t(_currentBuff[i] * 127.0);
+ itarget[i * 2 + 1] = 0;
+ }
+ }
+ else if (cSetup == FORMAT_STEREO_IQ) {
+ for (size_t i = 0; i < returnedElems; i++)
+ {
+ itarget[i * 2] = int8_t(_currentBuff[i * 2] * 127.0);
+ itarget[i * 2 + 1] = int8_t(_currentBuff[i * 2 + 1] * 127.0);
+ }
+ }
+ else if (cSetup == FORMAT_STEREO_QI) {
+ for (size_t i = 0; i < returnedElems; i++)
+ {
+ itarget[i * 2] = int8_t(_currentBuff[i * 2 + 1] * 127.0);
+ itarget[i * 2 + 1] = int8_t(_currentBuff[i * 2] * 127.0);
+ }
+ }
+ }
+ }
+
+ //bump variables for next call into readStream
+ bufferedElems -= returnedElems;
+ _currentBuff += returnedElems * elementsPerSample;
+
+ //return number of elements written to buff0
+ if (bufferedElems != 0) flags |= SOAPY_SDR_MORE_FRAGMENTS;
+ else this->releaseReadBuffer(stream, _currentHandle);
+ return returnedElems;
+}
+
+/*******************************************************************
+ * Direct buffer access API
+ ******************************************************************/
+
+size_t SoapyAudio::getNumDirectAccessBuffers(SoapySDR::Stream *stream)
+{
+ return _buffs.size();
+}
+
+int SoapyAudio::getDirectAccessBufferAddrs(SoapySDR::Stream *stream, const size_t handle, void **buffs)
+{
+ buffs[0] = (void *)_buffs[handle].data();
+ return 0;
+}
+
+int SoapyAudio::acquireReadBuffer(
+ SoapySDR::Stream *stream,
+ size_t &handle,
+ const void **buffs,
+ int &flags,
+ long long &timeNs,
+ const long timeoutUs)
+{
+ std::unique_lock <std::mutex> lock(_buf_mutex);
+
+ //reset is issued by various settings
+ //to drain old data out of the queue
+ if (resetBuffer)
+ {
+ //drain all buffers from the fifo
+ _buf_head = (_buf_head + _buf_count) % numBuffers;
+ _buf_count = 0;
+ resetBuffer = false;
+ _overflowEvent = false;
+ }
+
+ //handle overflow from the rx callback thread
+ if (_overflowEvent)
+ {
+ //drain the old buffers from the fifo
+ _buf_head = (_buf_head + _buf_count) % numBuffers;
+ _buf_count = 0;
+ _overflowEvent = false;
+ SoapySDR::log(SOAPY_SDR_SSI, "O");
+ return SOAPY_SDR_OVERFLOW;
+ }
+
+ //wait for a buffer to become available
+ while (_buf_count == 0)
+ {
+ _buf_cond.wait_for(lock, std::chrono::microseconds(timeoutUs));
+ if (_buf_count == 0) return SOAPY_SDR_TIMEOUT;
+ }
+
+ //extract handle and buffer
+ handle = _buf_head;
+ _buf_head = (_buf_head + 1) % numBuffers;
+ buffs[0] = (void *)_buffs[handle].data();
+ flags = 0;
+
+ //return number available
+ return _buffs[handle].size() / elementsPerSample;
+}
+
+void SoapyAudio::releaseReadBuffer(
+ SoapySDR::Stream *stream,
+ const size_t handle)
+{
+ //TODO this wont handle out of order releases
+ std::unique_lock <std::mutex> lock(_buf_mutex);
+ _buf_count--;
+}
diff --git a/debian/changelog b/debian/changelog
new file mode 100644
index 0000000..f7e4023
--- /dev/null
+++ b/debian/changelog
@@ -0,0 +1,5 @@
+soapyaudio (0.0.0) unstable; urgency=low
+
+ * pending
+
+ -- Josh Blum <josh at pothosware.com> Sun, 03 Jan 2016 13:18:35 -0800
diff --git a/debian/compat b/debian/compat
new file mode 100644
index 0000000..ec63514
--- /dev/null
+++ b/debian/compat
@@ -0,0 +1 @@
+9
diff --git a/debian/control b/debian/control
new file mode 100644
index 0000000..149e883
--- /dev/null
+++ b/debian/control
@@ -0,0 +1,22 @@
+Source: soapyaudio
+Section: libs
+Priority: optional
+Maintainer: Charles J. Cliffe <cj at cubicproductions.com>
+Uploaders: Josh Blum <josh at pothosware.com>
+Build-Depends:
+ debhelper (>= 9.0.0),
+ cmake,
+ libhamlib-dev,
+ libsoapysdr-dev
+Standards-Version: 3.9.5
+Homepage: https://github.com/pothosware/SoapyAudio/wiki
+Vcs-Git: https://github.com/pothosware/SoapyAudio.git
+Vcs-Browser: https://github.com/pothosware/SoapyAudio
+
+Package: soapysdr-audio
+Section: libs
+Architecture: any
+Pre-Depends: ${misc:Pre-Depends}
+Depends: ${shlibs:Depends}, ${misc:Depends}
+Description: Soapy Audio - Audio device support for Soapy SDR.
+ A Soapy module that supports Audio devices within the Soapy API.
diff --git a/debian/copyright b/debian/copyright
new file mode 100644
index 0000000..c561c81
--- /dev/null
+++ b/debian/copyright
@@ -0,0 +1,50 @@
+Format: http://www.debian.org/doc/packaging-manuals/copyright-format/1.0/
+Upstream-Name: soapyaudio
+Source: https://github.com/pothosware/SoapyAudio/wiki
+
+Files: *
+Copyright: Copyright (c) 2015 Charles J. Cliffe
+License: MIT
+ Permission is hereby granted, free of charge, to any person obtaining a copy
+ of this software and associated documentation files (the "Software"), to deal
+ in the Software without restriction, including without limitation the rights
+ to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ copies of the Software, and to permit persons to whom the Software is
+ furnished to do so, subject to the following conditions:
+ .
+ The above copyright notice and this permission notice shall be included in
+ all copies or substantial portions of the Software.
+ .
+ THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ THE SOFTWARE.
+
+Files: RtAudio/*
+Copyright: Copyright (c) 2001-2014 Gary P. Scavone
+License: The RtAudio license is similar to the MIT License.
+ Permission is hereby granted, free of charge, to any person
+ obtaining a copy of this software and associated documentation files
+ (the "Software"), to deal in the Software without restriction,
+ including without limitation the rights to use, copy, modify, merge,
+ publish, distribute, sublicense, and/or sell copies of the Software,
+ and to permit persons to whom the Software is furnished to do so,
+ subject to the following conditions:
+ .
+ The above copyright notice and this permission notice shall be
+ included in all copies or substantial portions of the Software.
+ .
+ Any person wishing to distribute modifications to the Software is
+ asked to send the modifications to the original developer so that
+ they can be incorporated into the canonical version. This is,
+ however, not a binding provision of this license.
+ .
+ THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+ EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+ ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+ CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
diff --git a/debian/docs b/debian/docs
new file mode 100644
index 0000000..b43bf86
--- /dev/null
+++ b/debian/docs
@@ -0,0 +1 @@
+README.md
diff --git a/debian/rules b/debian/rules
new file mode 100644
index 0000000..a904ad9
--- /dev/null
+++ b/debian/rules
@@ -0,0 +1,20 @@
+#!/usr/bin/make -f
+# -*- makefile -*-
+
+DEB_HOST_MULTIARCH ?= $(shell dpkg-architecture -qDEB_HOST_MULTIARCH)
+export DEB_HOST_MULTIARCH
+
+# Uncomment this to turn on verbose mode.
+#export DH_VERBOSE=1
+
+# This has to be exported to make some magic below work.
+export DH_OPTIONS
+
+
+%:
+ dh $@ --buildsystem=cmake --parallel
+
+override_dh_auto_configure:
+ dh_auto_configure -- \
+ -DLIB_SUFFIX="/$(DEB_HOST_MULTIARCH)" \
+ -DUSE_HAMLIB=ON
diff --git a/debian/source/format b/debian/source/format
new file mode 100644
index 0000000..163aaf8
--- /dev/null
+++ b/debian/source/format
@@ -0,0 +1 @@
+3.0 (quilt)
--
Alioth's /usr/local/bin/git-commit-notice on /srv/git.debian.org/git/pkg-hamradio/soapyaudio.git
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