Convert the examples from SDL_sound to libsndfile

9 files changed, 311 insertions, 880 deletions

diff --git a/CMakeLists.txt b/CMakeLists.txt
index 8fc2002f..92769b6c 100644
--- a/CMakeLists.txt
+++ b/CMakeLists.txt
@@ -1132,12 +1132,9 @@ IF(ALSOFT_UTILS AND NOT ALSOFT_NO_CONFIG_UTIL)
     add_subdirectory(utils/alsoft-config)
 ENDIF()
 IF(ALSOFT_EXAMPLES)
-    IF(NOT SDL2_FOUND)
-        FIND_PACKAGE(SDL2)
-    ENDIF()
+    FIND_PACKAGE(SndFile)
+    FIND_PACKAGE(SDL2)
     IF(SDL2_FOUND)
-        FIND_PACKAGE(SDL_sound)
-        FIND_PACKAGE(SndFile)
         FIND_PACKAGE(FFmpeg COMPONENTS AVFORMAT AVCODEC AVUTIL SWSCALE SWRESAMPLE)
     ENDIF()
 ENDIF()
@@ -1484,8 +1481,34 @@ IF(ALSOFT_EXAMPLES)
         TARGET_INCLUDE_DIRECTORIES(alplay PRIVATE ${SNDFILE_INCLUDE_DIRS})
         TARGET_LINK_LIBRARIES(alplay PRIVATE ${LINKER_FLAGS} ${SNDFILE_LIBRARIES} ex-common)
 
+        ADD_EXECUTABLE(alstream examples/alstream.c)
+        TARGET_INCLUDE_DIRECTORIES(alstream PRIVATE ${SNDFILE_INCLUDE_DIRS})
+        TARGET_LINK_LIBRARIES(alstream PRIVATE ${LINKER_FLAGS} ${SNDFILE_LIBRARIES} ex-common)
+
+        ADD_EXECUTABLE(alreverb examples/alreverb.c)
+        TARGET_INCLUDE_DIRECTORIES(alreverb PRIVATE ${SNDFILE_INCLUDE_DIRS})
+        TARGET_LINK_LIBRARIES(alreverb PRIVATE ${LINKER_FLAGS} ${SNDFILE_LIBRARIES} ex-common)
+
+        ADD_EXECUTABLE(almultireverb examples/almultireverb.c)
+        TARGET_INCLUDE_DIRECTORIES(almultireverb PRIVATE ${SNDFILE_INCLUDE_DIRS})
+        TARGET_LINK_LIBRARIES(almultireverb
+            PRIVATE ${LINKER_FLAGS} ${SNDFILE_LIBRARIES} ex-common ${MATH_LIB})
+
+        ADD_EXECUTABLE(allatency examples/allatency.c)
+        TARGET_INCLUDE_DIRECTORIES(allatency PRIVATE ${SNDFILE_INCLUDE_DIRS})
+        TARGET_LINK_LIBRARIES(allatency PRIVATE ${LINKER_FLAGS} ${SNDFILE_LIBRARIES} ex-common)
+
+        ADD_EXECUTABLE(alhrtf examples/alhrtf.c)
+        TARGET_INCLUDE_DIRECTORIES(alhrtf PRIVATE ${SNDFILE_INCLUDE_DIRS})
+        TARGET_LINK_LIBRARIES(alhrtf
+            PRIVATE ${LINKER_FLAGS} ${SNDFILE_LIBRARIES} ex-common ${MATH_LIB})
+
+        ADD_EXECUTABLE(alstreamcb examples/alstreamcb.cpp)
+        TARGET_INCLUDE_DIRECTORIES(alstreamcb PRIVATE ${SNDFILE_INCLUDE_DIRS})
+        TARGET_LINK_LIBRARIES(alstreamcb PRIVATE ${LINKER_FLAGS} ${SNDFILE_LIBRARIES} ex-common)
+
         IF(ALSOFT_INSTALL)
-            INSTALL(TARGETS alplay
+            INSTALL(TARGETS alplay alstream alreverb almultireverb allatency alhrtf
                 RUNTIME DESTINATION ${CMAKE_INSTALL_BINDIR}
                 LIBRARY DESTINATION ${CMAKE_INSTALL_LIBDIR}
                 ARCHIVE DESTINATION ${CMAKE_INSTALL_LIBDIR})
@@ -1495,60 +1518,20 @@ IF(ALSOFT_EXAMPLES)
     ENDIF()
 
     IF(SDL2_FOUND)
-        IF(SDL_SOUND_FOUND)
-            ADD_EXECUTABLE(alstream examples/alstream.c)
-            TARGET_INCLUDE_DIRECTORIES(alstream
-                PRIVATE ${SDL2_INCLUDE_DIR} ${SDL_SOUND_INCLUDE_DIR})
-            TARGET_LINK_LIBRARIES(alstream
-                PRIVATE ${LINKER_FLAGS} ${SDL_SOUND_LIBRARIES} ${SDL2_LIBRARY} ex-common)
-
-            ADD_EXECUTABLE(alstreamcb examples/alstreamcb.cpp)
-            TARGET_INCLUDE_DIRECTORIES(alstreamcb
-                PRIVATE ${SDL2_INCLUDE_DIR} ${SDL_SOUND_INCLUDE_DIR})
-            TARGET_LINK_LIBRARIES(alstreamcb
-                PRIVATE ${LINKER_FLAGS} ${SDL_SOUND_LIBRARIES} ${SDL2_LIBRARY} ex-common)
-
-            ADD_EXECUTABLE(alreverb examples/alreverb.c)
-            TARGET_INCLUDE_DIRECTORIES(alreverb
-                PRIVATE ${SDL2_INCLUDE_DIR} ${SDL_SOUND_INCLUDE_DIR})
-            TARGET_LINK_LIBRARIES(alreverb
-                PRIVATE ${LINKER_FLAGS} ${SDL_SOUND_LIBRARIES} ${SDL2_LIBRARY} ex-common)
-
-            ADD_EXECUTABLE(almultireverb examples/almultireverb.c)
-            TARGET_INCLUDE_DIRECTORIES(almultireverb
-                PRIVATE ${SDL2_INCLUDE_DIR} ${SDL_SOUND_INCLUDE_DIR})
-            TARGET_LINK_LIBRARIES(almultireverb
-                PRIVATE ${LINKER_FLAGS} ${SDL_SOUND_LIBRARIES} ${SDL2_LIBRARY} ex-common
-                ${MATH_LIB})
-
-            ADD_EXECUTABLE(allatency examples/allatency.c)
-            TARGET_INCLUDE_DIRECTORIES(allatency
-                PRIVATE ${SDL2_INCLUDE_DIR} ${SDL_SOUND_INCLUDE_DIR})
-            TARGET_LINK_LIBRARIES(allatency
-                PRIVATE ${LINKER_FLAGS} ${SDL_SOUND_LIBRARIES} ${SDL2_LIBRARY} ex-common)
-
-            ADD_EXECUTABLE(alloopback examples/alloopback.c)
-            TARGET_INCLUDE_DIRECTORIES(alloopback
-                PRIVATE ${SDL2_INCLUDE_DIR} ${SDL_SOUND_INCLUDE_DIR})
-            TARGET_LINK_LIBRARIES(alloopback
-                PRIVATE ${LINKER_FLAGS} ${SDL_SOUND_LIBRARIES} ${SDL2_LIBRARY} ex-common ${MATH_LIB})
-
-            ADD_EXECUTABLE(alhrtf examples/alhrtf.c)
-            TARGET_INCLUDE_DIRECTORIES(alhrtf
-                PRIVATE ${SDL2_INCLUDE_DIR} ${SDL_SOUND_INCLUDE_DIR})
-            TARGET_LINK_LIBRARIES(alhrtf
-                PRIVATE ${LINKER_FLAGS} ${SDL_SOUND_LIBRARIES} ${SDL2_LIBRARY} ex-common ${MATH_LIB})
-
-            IF(ALSOFT_INSTALL)
-                INSTALL(TARGETS alstream alreverb almultireverb allatency alloopback alhrtf
-                    RUNTIME DESTINATION ${CMAKE_INSTALL_BINDIR}
-                    LIBRARY DESTINATION ${CMAKE_INSTALL_LIBDIR}
-                    ARCHIVE DESTINATION ${CMAKE_INSTALL_LIBDIR})
-            ENDIF()
+        ADD_EXECUTABLE(alloopback examples/alloopback.c)
+        TARGET_INCLUDE_DIRECTORIES(alloopback PRIVATE ${SDL2_INCLUDE_DIR})
+        TARGET_LINK_LIBRARIES(alloopback
+            PRIVATE ${LINKER_FLAGS} ${SDL2_LIBRARY} ex-common ${MATH_LIB})
 
-            MESSAGE(STATUS "Building SDL_sound example programs")
+        IF(ALSOFT_INSTALL)
+            INSTALL(TARGETS alloopback
+                RUNTIME DESTINATION ${CMAKE_INSTALL_BINDIR}
+                LIBRARY DESTINATION ${CMAKE_INSTALL_LIBDIR}
+                ARCHIVE DESTINATION ${CMAKE_INSTALL_LIBDIR})
         ENDIF()
 
+        MESSAGE(STATUS "Building SDL example programs")
+
         SET(FFVER_OK FALSE)
         IF(FFMPEG_FOUND)
             SET(FFVER_OK TRUE)
diff --git a/cmake/FindSDL_sound.cmake b/cmake/FindSDL_sound.cmake
deleted file mode 100644
index 5557b55b..00000000
--- a/cmake/FindSDL_sound.cmake
+++ /dev/null
@@ -1,429 +0,0 @@
-# - Locates the SDL_sound library
-#
-# This module depends on SDL being found and
-# must be called AFTER FindSDL.cmake or FindSDL2.cmake is called.
-#
-# This module defines
-#  SDL_SOUND_INCLUDE_DIR, where to find SDL_sound.h
-#  SDL_SOUND_FOUND, if false, do not try to link to SDL_sound
-#  SDL_SOUND_LIBRARIES, this contains the list of libraries that you need
-#    to link against. This is a read-only variable and is marked INTERNAL.
-#  SDL_SOUND_EXTRAS, this is an optional variable for you to add your own
-#    flags to SDL_SOUND_LIBRARIES. This is prepended to SDL_SOUND_LIBRARIES.
-#    This is available mostly for cases this module failed to anticipate for
-#    and you must add additional flags. This is marked as ADVANCED.
-#  SDL_SOUND_VERSION_STRING, human-readable string containing the version of SDL_sound
-#
-# This module also defines (but you shouldn't need to use directly)
-#   SDL_SOUND_LIBRARY, the name of just the SDL_sound library you would link
-#   against. Use SDL_SOUND_LIBRARIES for you link instructions and not this one.
-# And might define the following as needed
-#   MIKMOD_LIBRARY
-#   MODPLUG_LIBRARY
-#   OGG_LIBRARY
-#   VORBIS_LIBRARY
-#   SMPEG_LIBRARY
-#   FLAC_LIBRARY
-#   SPEEX_LIBRARY
-#
-# Typically, you should not use these variables directly, and you should use
-# SDL_SOUND_LIBRARIES which contains SDL_SOUND_LIBRARY and the other audio libraries
-# (if needed) to successfully compile on your system.
-#
-# Created by Eric Wing.
-# This module is a bit more complicated than the other FindSDL* family modules.
-# The reason is that SDL_sound can be compiled in a large variety of different ways
-# which are independent of platform. SDL_sound may dynamically link against other 3rd
-# party libraries to get additional codec support, such as Ogg Vorbis, SMPEG, ModPlug,
-# MikMod, FLAC, Speex, and potentially others.
-# Under some circumstances which I don't fully understand,
-# there seems to be a requirement
-# that dependent libraries of libraries you use must also be explicitly
-# linked against in order to successfully compile. SDL_sound does not currently
-# have any system in place to know how it was compiled.
-# So this CMake module does the hard work in trying to discover which 3rd party
-# libraries are required for building (if any).
-# This module uses a brute force approach to create a test program that uses SDL_sound,
-# and then tries to build it. If the build fails, it parses the error output for
-# known symbol names to figure out which libraries are needed.
-#
-# Responds to the $SDLDIR and $SDLSOUNDDIR environmental variable that would
-# correspond to the ./configure --prefix=$SDLDIR used in building SDL.
-#
-# On OSX, this will prefer the Framework version (if found) over others.
-# People will have to manually change the cache values of
-# SDL_LIBRARY or SDL2_LIBRARY to override this selection or set the CMake
-# environment CMAKE_INCLUDE_PATH to modify the search paths.
-
-#=============================================================================
-# Copyright 2005-2009 Kitware, Inc.
-# Copyright 2012 Benjamin Eikel
-#
-# Distributed under the OSI-approved BSD License (the "License");
-# see accompanying file Copyright.txt for details.
-#
-# This software is distributed WITHOUT ANY WARRANTY; without even the
-# implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
-# See the License for more information.
-#=============================================================================
-# (To distribute this file outside of CMake, substitute the full
-#  License text for the above reference.)
-
-set(SDL_SOUND_EXTRAS "" CACHE STRING "SDL_sound extra flags")
-mark_as_advanced(SDL_SOUND_EXTRAS)
-
-# Find SDL_sound.h
-find_path(SDL_SOUND_INCLUDE_DIR SDL_sound.h
-    HINTS
-        ENV SDLSOUNDDIR
-        ENV SDLDIR
-    PATH_SUFFIXES SDL SDL12 SDL11
-)
-
-find_library(SDL_SOUND_LIBRARY
-    NAMES SDL_sound
-    HINTS
-        ENV SDLSOUNDDIR
-        ENV SDLDIR
-)
-
-if(SDL2_FOUND OR SDL_FOUND)
-    if(SDL_SOUND_INCLUDE_DIR AND SDL_SOUND_LIBRARY)
-        # CMake is giving me problems using TRY_COMPILE with the CMAKE_FLAGS
-        # for the :STRING syntax if I have multiple values contained in a
-        # single variable. This is a problem for the SDL2_LIBRARY variable
-        # because it does just that. When I feed this variable to the command,
-        # only the first value gets the appropriate modifier (e.g. -I) and
-        # the rest get dropped.
-        # To get multiple single variables to work, I must separate them with a "\;"
-        # I could go back and modify the FindSDL2.cmake module, but that's kind of painful.
-        # The solution would be to try something like:
-        # set(SDL2_TRY_COMPILE_LIBRARY_LIST "${SDL2_TRY_COMPILE_LIBRARY_LIST}\;${CMAKE_THREAD_LIBS_INIT}")
-        # Instead, it was suggested on the mailing list to write a temporary CMakeLists.txt
-        # with a temporary test project and invoke that with TRY_COMPILE.
-        # See message thread "Figuring out dependencies for a library in order to build"
-        # 2005-07-16
-        #     try_compile(
-        #             MY_RESULT
-        #             ${CMAKE_BINARY_DIR}
-        #             ${PROJECT_SOURCE_DIR}/DetermineSoundLibs.c
-        #             CMAKE_FLAGS
-        #                     -DINCLUDE_DIRECTORIES:STRING=${SDL2_INCLUDE_DIR}\;${SDL_SOUND_INCLUDE_DIR}
-        #                     -DLINK_LIBRARIES:STRING=${SDL_SOUND_LIBRARY}\;${SDL2_LIBRARY}
-        #             OUTPUT_VARIABLE MY_OUTPUT
-        #     )
-
-        # To minimize external dependencies, create a sdlsound test program
-        # which will be used to figure out if additional link dependencies are
-        # required for the link phase.
-        file(WRITE ${PROJECT_BINARY_DIR}/CMakeTmp/DetermineSoundLibs.c
-            "#include \"SDL_sound.h\"
-            #include \"SDL.h\"
-            int main(int argc, char* argv[])
-            {
-                Sound_AudioInfo desired;
-                Sound_Sample* sample;
-
-                SDL_Init(0);
-                Sound_Init();
-
-                /* This doesn't actually have to work, but Init() is a no-op
-                 * for some of the decoders, so this should force more symbols
-                 * to be pulled in.
-                 */
-                sample = Sound_NewSampleFromFile(argv[1], &desired, 4096);
-
-                Sound_Quit();
-                SDL_Quit();
-                return 0;
-            }"
-        )
-
-        # Calling
-        # target_link_libraries(DetermineSoundLibs "${SDL_SOUND_LIBRARY} ${SDL2_LIBRARY})
-        # causes problems when SDL2_LIBRARY looks like
-        # /Library/Frameworks/SDL2.framework;-framework Cocoa
-        # The ;-framework Cocoa seems to be confusing CMake once the OS X
-        # framework support was added. I was told that breaking up the list
-        # would fix the problem.
-        set(TMP_LIBS "")
-        if(SDL2_FOUND)
-            set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARY} ${SDL2_LIBRARY})
-            foreach(lib ${SDL_SOUND_LIBRARY} ${SDL2_LIBRARY})
-                set(TMP_LIBS "${TMP_LIBS} \"${lib}\"")
-            endforeach()
-            set(TMP_INCLUDE_DIRS ${SDL2_INCLUDE_DIR} ${SDL_SOUND_INCLUDE_DIR})
-        else()
-            set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARY} ${SDL_LIBRARY})
-            foreach(lib ${SDL_SOUND_LIBRARY} ${SDL_LIBRARY})
-                set(TMP_LIBS "${TMP_LIBS} \"${lib}\"")
-            endforeach()
-            set(TMP_INCLUDE_DIRS ${SDL_INCLUDE_DIR} ${SDL_SOUND_INCLUDE_DIR})
-        endif()
-
-        # Keep trying to build a temp project until we find all missing libs.
-        set(TRY_AGAIN TRUE)
-        WHILE(TRY_AGAIN)
-            set(TRY_AGAIN FALSE)
-            # message("TMP_TRY_LIBS ${TMP_TRY_LIBS}")
-
-            # Write the CMakeLists.txt and test project
-            # Weird, this is still sketchy. If I don't quote the variables
-            # in the TARGET_LINK_LIBRARIES, I seem to loose everything
-            # in the SDL2_LIBRARY string after the "-framework".
-            # But if I quote the stuff in INCLUDE_DIRECTORIES, it doesn't work.
-            file(WRITE ${PROJECT_BINARY_DIR}/CMakeTmp/CMakeLists.txt
-                "cmake_minimum_required(VERSION 2.8)
-                project(DetermineSoundLibs C)
-                include_directories(${TMP_INCLUDE_DIRS})
-                add_executable(DetermineSoundLibs DetermineSoundLibs.c)
-                target_link_libraries(DetermineSoundLibs ${TMP_LIBS})"
-            )
-
-            try_compile(
-                MY_RESULT
-                ${PROJECT_BINARY_DIR}/CMakeTmp
-                ${PROJECT_BINARY_DIR}/CMakeTmp
-                DetermineSoundLibs
-                OUTPUT_VARIABLE MY_OUTPUT
-            )
-            # message("${MY_RESULT}")
-            # message(${MY_OUTPUT})
-
-            if(NOT MY_RESULT)
-                # I expect that MPGLIB, VOC, WAV, AIFF, and SHN are compiled in statically.
-                # I think Timidity is also compiled in statically.
-                # I've never had to explcitly link against Quicktime, so I'll skip that for now.
-
-                # Find libmath
-                if("${MY_OUTPUT}" MATCHES "cos@@GLIBC")
-                    find_library(MATH_LIBRARY NAMES m)
-                    if(MATH_LIBRARY)
-                        set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARIES_TMP} ${MATH_LIBRARY})
-                        set(TMP_LIBS "${SDL_SOUND_LIBRARIES_TMP} \"${MATH_LIBRARY}\"")
-                        set(TRY_AGAIN TRUE)
-                    endif(MATH_LIBRARY)
-                endif("${MY_OUTPUT}" MATCHES "cos@@GLIBC")
-
-                # Find MikMod
-                if("${MY_OUTPUT}" MATCHES "MikMod_")
-                    find_library(MIKMOD_LIBRARY
-                        NAMES libmikmod-coreaudio mikmod
-                        PATHS
-                            ENV MIKMODDIR
-                            ENV SDLSOUNDDIR
-                            ENV SDLDIR
-                            /sw
-                            /opt/local
-                            /opt/csw
-                            /opt
-                        PATH_SUFFIXES lib
-                    )
-                    if(MIKMOD_LIBRARY)
-                        set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARIES_TMP} ${MIKMOD_LIBRARY})
-                        set(TMP_LIBS "${SDL_SOUND_LIBRARIES_TMP} \"${MIKMOD_LIBRARY}\"")
-                        set(TRY_AGAIN TRUE)
-                    endif(MIKMOD_LIBRARY)
-                endif("${MY_OUTPUT}" MATCHES "MikMod_")
-
-                # Find ModPlug
-                if("${MY_OUTPUT}" MATCHES "MODPLUG_")
-                    find_library(MODPLUG_LIBRARY
-                        NAMES modplug
-                        PATHS
-                            ENV MODPLUGDIR
-                            ENV SDLSOUNDDIR
-                            ENV SDLDIR
-                            /sw
-                            /opt/local
-                            /opt/csw
-                            /opt
-                        PATH_SUFFIXES lib
-                    )
-                    if(MODPLUG_LIBRARY)
-                        set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARIES_TMP} ${MODPLUG_LIBRARY})
-                        set(TMP_LIBS "${SDL_SOUND_LIBRARIES_TMP} \"${MODPLUG_LIBRARY}\"")
-                        set(TRY_AGAIN TRUE)
-                    endif()
-                endif()
-
-                # Find Ogg and Vorbis
-                if("${MY_OUTPUT}" MATCHES "ov_")
-                    find_library(VORBISFILE_LIBRARY
-                        NAMES vorbisfile VorbisFile VORBISFILE
-                        PATHS
-                            ENV VORBISDIR
-                            ENV OGGDIR
-                            ENV SDLSOUNDDIR
-                            ENV SDLDIR
-                            /sw
-                            /opt/local
-                            /opt/csw
-                            /opt
-                        PATH_SUFFIXES lib
-                    )
-                    if(VORBISFILE_LIBRARY)
-                        set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARIES_TMP} ${VORBISFILE_LIBRARY})
-                        set(TMP_LIBS "${SDL_SOUND_LIBRARIES_TMP} \"${VORBISFILE_LIBRARY}\"")
-                        set(TRY_AGAIN TRUE)
-                    endif()
-
-                    find_library(VORBIS_LIBRARY
-                        NAMES vorbis Vorbis VORBIS
-                        PATHS
-                            ENV OGGDIR
-                            ENV VORBISDIR
-                            ENV SDLSOUNDDIR
-                            ENV SDLDIR
-                            /sw
-                            /opt/local
-                            /opt/csw
-                            /opt
-                        PATH_SUFFIXES lib
-                    )
-                    if(VORBIS_LIBRARY)
-                        set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARIES_TMP} ${VORBIS_LIBRARY})
-                        set(TMP_LIBS "${SDL_SOUND_LIBRARIES_TMP} \"${VORBIS_LIBRARY}\"")
-                        set(TRY_AGAIN TRUE)
-                    endif()
-
-                    find_library(OGG_LIBRARY
-                        NAMES ogg Ogg OGG
-                        PATHS
-                            ENV OGGDIR
-                            ENV VORBISDIR
-                            ENV SDLSOUNDDIR
-                            ENV SDLDIR
-                            /sw
-                            /opt/local
-                            /opt/csw
-                            /opt
-                        PATH_SUFFIXES lib
-                    )
-                    if(OGG_LIBRARY)
-                        set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARIES_TMP} ${OGG_LIBRARY})
-                        set(TMP_LIBS "${SDL_SOUND_LIBRARIES_TMP} \"${OGG_LIBRARY}\"")
-                        set(TRY_AGAIN TRUE)
-                    endif()
-                endif()
-
-                # Find SMPEG
-                if("${MY_OUTPUT}" MATCHES "SMPEG_")
-                    find_library(SMPEG_LIBRARY
-                        NAMES smpeg SMPEG Smpeg SMpeg
-                        PATHS
-                            ENV SMPEGDIR
-                            ENV SDLSOUNDDIR
-                            ENV SDLDIR
-                            /sw
-                            /opt/local
-                            /opt/csw
-                            /opt
-                        PATH_SUFFIXES lib
-                    )
-                    if(SMPEG_LIBRARY)
-                        set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARIES_TMP} ${SMPEG_LIBRARY})
-                        set(TMP_LIBS "${SDL_SOUND_LIBRARIES_TMP} \"${SMPEG_LIBRARY}\"")
-                        set(TRY_AGAIN TRUE)
-                    endif()
-                endif()
-
-
-                # Find FLAC
-                if("${MY_OUTPUT}" MATCHES "FLAC_")
-                    find_library(FLAC_LIBRARY
-                        NAMES flac FLAC
-                        PATHS
-                            ENV FLACDIR
-                            ENV SDLSOUNDDIR
-                            ENV SDLDIR
-                            /sw
-                            /opt/local
-                            /opt/csw
-                            /opt
-                        PATH_SUFFIXES lib
-                    )
-                    if(FLAC_LIBRARY)
-                        set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARIES_TMP} ${FLAC_LIBRARY})
-                        set(TMP_LIBS "${SDL_SOUND_LIBRARIES_TMP} \"${FLAC_LIBRARY}\"")
-                        set(TRY_AGAIN TRUE)
-                    endif()
-                endif()
-
-
-                # Hmmm...Speex seems to depend on Ogg. This might be a problem if
-                # the TRY_COMPILE attempt gets blocked at SPEEX before it can pull
-                # in the Ogg symbols. I'm not sure if I should duplicate the ogg stuff
-                # above for here or if two ogg entries will screw up things.
-                if("${MY_OUTPUT}" MATCHES "speex_")
-                    find_library(SPEEX_LIBRARY
-                        NAMES speex SPEEX
-                        PATHS
-                            ENV SPEEXDIR
-                            ENV SDLSOUNDDIR
-                            ENV SDLDIR
-                            /sw
-                            /opt/local
-                            /opt/csw
-                            /opt
-                        PATH_SUFFIXES lib
-                    )
-                    if(SPEEX_LIBRARY)
-                        set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARIES_TMP} ${SPEEX_LIBRARY})
-                        set(TMP_LIBS "${SDL_SOUND_LIBRARIES_TMP} \"${SPEEX_LIBRARY}\"")
-                        set(TRY_AGAIN TRUE)
-                    endif()
-
-                    # Find OGG (needed for Speex)
-                    # We might have already found Ogg for Vorbis, so skip it if so.
-                    if(NOT OGG_LIBRARY)
-                        find_library(OGG_LIBRARY
-                            NAMES ogg Ogg OGG
-                            PATHS
-                                ENV OGGDIR
-                                ENV VORBISDIR
-                                ENV SPEEXDIR
-                                ENV SDLSOUNDDIR
-                                ENV SDLDIR
-                                /sw
-                                /opt/local
-                                /opt/csw
-                                /opt
-                            PATH_SUFFIXES lib
-                        )
-                        if(OGG_LIBRARY)
-                            set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARIES_TMP} ${OGG_LIBRARY})
-                            set(TMP_LIBS "${SDL_SOUND_LIBRARIES_TMP} \"${OGG_LIBRARY}\"")
-                            set(TRY_AGAIN TRUE)
-                        endif()
-                    endif()
-                endif()
-            endif()
-        ENDWHILE()
-        unset(TMP_INCLUDE_DIRS)
-        unset(TMP_LIBS)
-
-        set(SDL_SOUND_LIBRARIES ${SDL_SOUND_EXTRAS} ${SDL_SOUND_LIBRARIES_TMP} CACHE INTERNAL "SDL_sound and dependent libraries")
-    endif()
-endif()
-
-if(SDL_SOUND_INCLUDE_DIR AND EXISTS "${SDL_SOUND_INCLUDE_DIR}/SDL_sound.h")
-  file(STRINGS "${SDL_SOUND_INCLUDE_DIR}/SDL_sound.h" SDL_SOUND_VERSION_MAJOR_LINE REGEX "^#define[ \t]+SOUND_VER_MAJOR[ \t]+[0-9]+$")
-  file(STRINGS "${SDL_SOUND_INCLUDE_DIR}/SDL_sound.h" SDL_SOUND_VERSION_MINOR_LINE REGEX "^#define[ \t]+SOUND_VER_MINOR[ \t]+[0-9]+$")
-  file(STRINGS "${SDL_SOUND_INCLUDE_DIR}/SDL_sound.h" SDL_SOUND_VERSION_PATCH_LINE REGEX "^#define[ \t]+SOUND_VER_PATCH[ \t]+[0-9]+$")
-  string(REGEX REPLACE "^#define[ \t]+SOUND_VER_MAJOR[ \t]+([0-9]+)$" "\\1" SDL_SOUND_VERSION_MAJOR "${SDL_SOUND_VERSION_MAJOR_LINE}")
-  string(REGEX REPLACE "^#define[ \t]+SOUND_VER_MINOR[ \t]+([0-9]+)$" "\\1" SDL_SOUND_VERSION_MINOR "${SDL_SOUND_VERSION_MINOR_LINE}")
-  string(REGEX REPLACE "^#define[ \t]+SOUND_VER_PATCH[ \t]+([0-9]+)$" "\\1" SDL_SOUND_VERSION_PATCH "${SDL_SOUND_VERSION_PATCH_LINE}")
-  set(SDL_SOUND_VERSION_STRING ${SDL_SOUND_VERSION_MAJOR}.${SDL_SOUND_VERSION_MINOR}.${SDL_SOUND_VERSION_PATCH})
-  unset(SDL_SOUND_VERSION_MAJOR_LINE)
-  unset(SDL_SOUND_VERSION_MINOR_LINE)
-  unset(SDL_SOUND_VERSION_PATCH_LINE)
-  unset(SDL_SOUND_VERSION_MAJOR)
-  unset(SDL_SOUND_VERSION_MINOR)
-  unset(SDL_SOUND_VERSION_PATCH)
-endif()
-
-include(FindPackageHandleStandardArgs)
-FIND_PACKAGE_HANDLE_STANDARD_ARGS(SDL_sound
-                                  REQUIRED_VARS SDL_SOUND_LIBRARIES SDL_SOUND_INCLUDE_DIR
-                                  VERSION_VAR SDL_SOUND_VERSION_STRING)
diff --git a/examples/alhrtf.c b/examples/alhrtf.c
index 2be28a91..bea87f56 100644
--- a/examples/alhrtf.c
+++ b/examples/alhrtf.c
@@ -25,13 +25,14 @@
 /* This file contains an example for selecting an HRTF. */
 
 #include <assert.h>
+#include <inttypes.h>
+#include <limits.h>
 #include <math.h>
 #include <stdio.h>
+#include <stdlib.h>
 #include <string.h>
 
-#include "SDL_sound.h"
-#include "SDL_audio.h"
-#include "SDL_stdinc.h"
+#include "sndfile.h"
 
 #include "AL/al.h"
 #include "AL/alc.h"
@@ -52,68 +53,62 @@ static LPALCRESETDEVICESOFT alcResetDeviceSOFT;
  */
 static ALuint LoadSound(const char *filename)
 {
-    Sound_Sample *sample;
     ALenum err, format;
     ALuint buffer;
-    Uint32 slen;
-
-    /* Open the audio file */
-    sample = Sound_NewSampleFromFile(filename, NULL, 65536);
-    if(!sample)
+    SNDFILE *sndfile;
+    SF_INFO sfinfo;
+    short *membuf;
+    sf_count_t num_frames;
+    ALsizei num_bytes;
+
+    /* Open the audio file and check that it's usable. */
+    sndfile = sf_open(filename, SFM_READ, &sfinfo);
+    if(!sndfile)
     {
-        fprintf(stderr, "Could not open audio in %s\n", filename);
+        fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(sndfile));
         return 0;
     }
-
-    /* Get the sound format, and figure out the OpenAL format */
-    if(sample->actual.channels == 1)
-    {
-        if(sample->actual.format == AUDIO_U8)
-            format = AL_FORMAT_MONO8;
-        else if(sample->actual.format == AUDIO_S16SYS)
-            format = AL_FORMAT_MONO16;
-        else
-        {
-            fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format);
-            Sound_FreeSample(sample);
-            return 0;
-        }
-    }
-    else if(sample->actual.channels == 2)
+    if(sfinfo.frames < 1 || sfinfo.frames > (sf_count_t)(INT_MAX/sizeof(short))/sfinfo.channels)
     {
-        if(sample->actual.format == AUDIO_U8)
-            format = AL_FORMAT_STEREO8;
-        else if(sample->actual.format == AUDIO_S16SYS)
-            format = AL_FORMAT_STEREO16;
-        else
-        {
-            fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format);
-            Sound_FreeSample(sample);
-            return 0;
-        }
+        fprintf(stderr, "Bad sample count in %s (%" PRId64 ")\n", filename, sfinfo.frames);
+        sf_close(sndfile);
+        return 0;
     }
+
+    /* Get the sound format, and figure out the OpenAL format */
+    if(sfinfo.channels == 1)
+        format = AL_FORMAT_MONO16;
+    else if(sfinfo.channels == 2)
+        format = AL_FORMAT_STEREO16;
     else
     {
-        fprintf(stderr, "Unsupported channel count: %d\n", sample->actual.channels);
-        Sound_FreeSample(sample);
+        fprintf(stderr, "Unsupported channel count: %d\n", sfinfo.channels);
+        sf_close(sndfile);
         return 0;
     }
 
-    /* Decode the whole audio stream to a buffer. */
-    slen = Sound_DecodeAll(sample);
-    if(!sample->buffer || slen == 0)
+    /* Decode the whole audio file to a buffer. */
+    membuf = malloc((size_t)(sfinfo.frames * sfinfo.channels) * sizeof(short));
+
+    num_frames = sf_readf_short(sndfile, membuf, sfinfo.frames);
+    if(num_frames < 1)
     {
-        fprintf(stderr, "Failed to read audio from %s\n", filename);
-        Sound_FreeSample(sample);
+        free(membuf);
+        sf_close(sndfile);
+        fprintf(stderr, "Failed to read samples in %s (%" PRId64 ")\n", filename, num_frames);
         return 0;
     }
+    num_bytes = (ALsizei)(num_frames * sfinfo.channels) * (ALsizei)sizeof(short);
 
     /* Buffer the audio data into a new buffer object, then free the data and
-     * close the file. */
+     * close the file.
+     */
     buffer = 0;
     alGenBuffers(1, &buffer);
-    alBufferData(buffer, format, sample->buffer, (ALsizei)slen, (ALsizei)sample->actual.rate);
-    Sound_FreeSample(sample);
+    alBufferData(buffer, format, membuf, num_bytes, sfinfo.samplerate);
+
+    free(membuf);
+    sf_close(sndfile);
 
     /* Check if an error occured, and clean up if so. */
     err = alGetError();
@@ -240,14 +235,10 @@ int main(int argc, char **argv)
     }
     fflush(stdout);
 
-    /* Initialize SDL_sound. */
-    Sound_Init();
-
     /* Load the sound into a buffer. */
     buffer = LoadSound(soundname);
     if(!buffer)
     {
-        Sound_Quit();
         CloseAL();
         return 1;
     }
@@ -291,11 +282,9 @@ int main(int argc, char **argv)
         alGetSourcei(source, AL_SOURCE_STATE, &state);
     } while(alGetError() == AL_NO_ERROR && state == AL_PLAYING);
 
-    /* All done. Delete resources, and close down SDL_sound and OpenAL. */
+    /* All done. Delete resources, and close down OpenAL. */
     alDeleteSources(1, &source);
     alDeleteBuffers(1, &buffer);
-
-    Sound_Quit();
     CloseAL();
 
     return 0;
diff --git a/examples/allatency.c b/examples/allatency.c
index a61fb820..7ece01eb 100644
--- a/examples/allatency.c
+++ b/examples/allatency.c
@@ -24,12 +24,13 @@
 
 /* This file contains an example for checking the latency of a sound. */
 
-#include <stdio.h>
 #include <assert.h>
+#include <inttypes.h>
+#include <limits.h>
+#include <stdio.h>
+#include <stdlib.h>
 
-#include "SDL_sound.h"
-#include "SDL_audio.h"
-#include "SDL_stdinc.h"
+#include "sndfile.h"
 
 #include "AL/al.h"
 #include "AL/alext.h"
@@ -55,68 +56,62 @@ static LPALGETSOURCEI64VSOFT alGetSourcei64vSOFT;
  */
 static ALuint LoadSound(const char *filename)
 {
-    Sound_Sample *sample;
     ALenum err, format;
     ALuint buffer;
-    Uint32 slen;
-
-    /* Open the audio file */
-    sample = Sound_NewSampleFromFile(filename, NULL, 65536);
-    if(!sample)
+    SNDFILE *sndfile;
+    SF_INFO sfinfo;
+    short *membuf;
+    sf_count_t num_frames;
+    ALsizei num_bytes;
+
+    /* Open the audio file and check that it's usable. */
+    sndfile = sf_open(filename, SFM_READ, &sfinfo);
+    if(!sndfile)
     {
-        fprintf(stderr, "Could not open audio in %s\n", filename);
+        fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(sndfile));
         return 0;
     }
-
-    /* Get the sound format, and figure out the OpenAL format */
-    if(sample->actual.channels == 1)
-    {
-        if(sample->actual.format == AUDIO_U8)
-            format = AL_FORMAT_MONO8;
-        else if(sample->actual.format == AUDIO_S16SYS)
-            format = AL_FORMAT_MONO16;
-        else
-        {
-            fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format);
-            Sound_FreeSample(sample);
-            return 0;
-        }
-    }
-    else if(sample->actual.channels == 2)
+    if(sfinfo.frames < 1 || sfinfo.frames > (sf_count_t)(INT_MAX/sizeof(short))/sfinfo.channels)
     {
-        if(sample->actual.format == AUDIO_U8)
-            format = AL_FORMAT_STEREO8;
-        else if(sample->actual.format == AUDIO_S16SYS)
-            format = AL_FORMAT_STEREO16;
-        else
-        {
-            fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format);
-            Sound_FreeSample(sample);
-            return 0;
-        }
+        fprintf(stderr, "Bad sample count in %s (%" PRId64 ")\n", filename, sfinfo.frames);
+        sf_close(sndfile);
+        return 0;
     }
+
+    /* Get the sound format, and figure out the OpenAL format */
+    if(sfinfo.channels == 1)
+        format = AL_FORMAT_MONO16;
+    else if(sfinfo.channels == 2)
+        format = AL_FORMAT_STEREO16;
     else
     {
-        fprintf(stderr, "Unsupported channel count: %d\n", sample->actual.channels);
-        Sound_FreeSample(sample);
+        fprintf(stderr, "Unsupported channel count: %d\n", sfinfo.channels);
+        sf_close(sndfile);
         return 0;
     }
 
-    /* Decode the whole audio stream to a buffer. */
-    slen = Sound_DecodeAll(sample);
-    if(!sample->buffer || slen == 0)
+    /* Decode the whole audio file to a buffer. */
+    membuf = malloc((size_t)(sfinfo.frames * sfinfo.channels) * sizeof(short));
+
+    num_frames = sf_readf_short(sndfile, membuf, sfinfo.frames);
+    if(num_frames < 1)
     {
-        fprintf(stderr, "Failed to read audio from %s\n", filename);
-        Sound_FreeSample(sample);
+        free(membuf);
+        sf_close(sndfile);
+        fprintf(stderr, "Failed to read samples in %s (%" PRId64 ")\n", filename, num_frames);
         return 0;
     }
+    num_bytes = (ALsizei)(num_frames * sfinfo.channels) * (ALsizei)sizeof(short);
 
     /* Buffer the audio data into a new buffer object, then free the data and
-     * close the file. */
+     * close the file.
+     */
     buffer = 0;
     alGenBuffers(1, &buffer);
-    alBufferData(buffer, format, sample->buffer, (ALsizei)slen, (ALsizei)sample->actual.rate);
-    Sound_FreeSample(sample);
+    alBufferData(buffer, format, membuf, num_bytes, sfinfo.samplerate);
+
+    free(membuf);
+    sf_close(sndfile);
 
     /* Check if an error occured, and clean up if so. */
     err = alGetError();
@@ -173,14 +168,10 @@ int main(int argc, char **argv)
     LOAD_PROC(LPALGETSOURCEI64VSOFT, alGetSourcei64vSOFT);
 #undef LOAD_PROC
 
-    /* Initialize SDL_sound. */
-    Sound_Init();
-
     /* Load the sound into a buffer. */
     buffer = LoadSound(argv[0]);
     if(!buffer)
     {
-        Sound_Quit();
         CloseAL();
         return 1;
     }
@@ -206,11 +197,9 @@ int main(int argc, char **argv)
     } while(alGetError() == AL_NO_ERROR && state == AL_PLAYING);
     printf("\n");
 
-    /* All done. Delete resources, and close down SDL_sound and OpenAL. */
+    /* All done. Delete resources, and close down OpenAL. */
     alDeleteSources(1, &source);
     alDeleteBuffers(1, &buffer);
-
-    Sound_Quit();
     CloseAL();
 
     return 0;
diff --git a/examples/almultireverb.c b/examples/almultireverb.c
index a90b3368..eb874061 100644
--- a/examples/almultireverb.c
+++ b/examples/almultireverb.c
@@ -29,14 +29,16 @@
  * listener.
  */
 
+
 #include <assert.h>
+#include <inttypes.h>
+#include <limits.h>
 #include <math.h>
 #include <stdio.h>
+#include <stdlib.h>
 #include <string.h>
 
-#include "SDL_sound.h"
-#include "SDL_audio.h"
-#include "SDL_stdinc.h"
+#include "sndfile.h"
 
 #include "AL/al.h"
 #include "AL/alc.h"
@@ -151,68 +153,62 @@ static int LoadEffect(ALuint effect, const EFXEAXREVERBPROPERTIES *reverb)
  */
 static ALuint LoadSound(const char *filename)
 {
-    Sound_Sample *sample;
     ALenum err, format;
     ALuint buffer;
-    Uint32 slen;
-
-    /* Open the audio file */
-    sample = Sound_NewSampleFromFile(filename, NULL, 65536);
-    if(!sample)
+    SNDFILE *sndfile;
+    SF_INFO sfinfo;
+    short *membuf;
+    sf_count_t num_frames;
+    ALsizei num_bytes;
+
+    /* Open the audio file and check that it's usable. */
+    sndfile = sf_open(filename, SFM_READ, &sfinfo);
+    if(!sndfile)
     {
-        fprintf(stderr, "Could not open audio in %s\n", filename);
+        fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(sndfile));
         return 0;
     }
-
-    /* Get the sound format, and figure out the OpenAL format */
-    if(sample->actual.channels == 1)
-    {
-        if(sample->actual.format == AUDIO_U8)
-            format = AL_FORMAT_MONO8;
-        else if(sample->actual.format == AUDIO_S16SYS)
-            format = AL_FORMAT_MONO16;
-        else
-        {
-            fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format);
-            Sound_FreeSample(sample);
-            return 0;
-        }
-    }
-    else if(sample->actual.channels == 2)
+    if(sfinfo.frames < 1 || sfinfo.frames > (sf_count_t)(INT_MAX/sizeof(short))/sfinfo.channels)
     {
-        if(sample->actual.format == AUDIO_U8)
-            format = AL_FORMAT_STEREO8;
-        else if(sample->actual.format == AUDIO_S16SYS)
-            format = AL_FORMAT_STEREO16;
-        else
-        {
-            fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format);
-            Sound_FreeSample(sample);
-            return 0;
-        }
+        fprintf(stderr, "Bad sample count in %s (%" PRId64 ")\n", filename, sfinfo.frames);
+        sf_close(sndfile);
+        return 0;
     }
+
+    /* Get the sound format, and figure out the OpenAL format */
+    if(sfinfo.channels == 1)
+        format = AL_FORMAT_MONO16;
+    else if(sfinfo.channels == 2)
+        format = AL_FORMAT_STEREO16;
     else
     {
-        fprintf(stderr, "Unsupported channel count: %d\n", sample->actual.channels);
-        Sound_FreeSample(sample);
+        fprintf(stderr, "Unsupported channel count: %d\n", sfinfo.channels);
+        sf_close(sndfile);
         return 0;
     }
 
-    /* Decode the whole audio stream to a buffer. */
-    slen = Sound_DecodeAll(sample);
-    if(!sample->buffer || slen == 0)
+    /* Decode the whole audio file to a buffer. */
+    membuf = malloc((size_t)(sfinfo.frames * sfinfo.channels) * sizeof(short));
+
+    num_frames = sf_readf_short(sndfile, membuf, sfinfo.frames);
+    if(num_frames < 1)
     {
-        fprintf(stderr, "Failed to read audio from %s\n", filename);
-        Sound_FreeSample(sample);
+        free(membuf);
+        sf_close(sndfile);
+        fprintf(stderr, "Failed to read samples in %s (%" PRId64 ")\n", filename, num_frames);
         return 0;
     }
+    num_bytes = (ALsizei)(num_frames * sfinfo.channels) * (ALsizei)sizeof(short);
 
     /* Buffer the audio data into a new buffer object, then free the data and
-     * close the file. */
+     * close the file.
+     */
     buffer = 0;
     alGenBuffers(1, &buffer);
-    alBufferData(buffer, format, sample->buffer, (ALsizei)slen, (ALsizei)sample->actual.rate);
-    Sound_FreeSample(sample);
+    alBufferData(buffer, format, membuf, num_bytes, sfinfo.samplerate);
+
+    free(membuf);
+    sf_close(sndfile);
 
     /* Check if an error occured, and clean up if so. */
     err = alGetError();
@@ -561,15 +557,11 @@ int main(int argc, char **argv)
     LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTFV, alGetAuxiliaryEffectSlotfv);
 #undef LOAD_PROC
 
-    /* Initialize SDL_sound. */
-    Sound_Init();
-
     /* Load the sound into a buffer. */
     buffer = LoadSound(argv[0]);
     if(!buffer)
     {
         CloseAL();
-        Sound_Quit();
         return 1;
     }
 
@@ -585,7 +577,6 @@ int main(int argc, char **argv)
     {
         alDeleteEffects(2, effects);
         alDeleteBuffers(1, &buffer);
-        Sound_Quit();
         CloseAL();
         return 1;
     }
@@ -684,14 +675,13 @@ int main(int argc, char **argv)
         alGetSourcei(source, AL_SOURCE_STATE, &state);
     } while(alGetError() == AL_NO_ERROR && state == AL_PLAYING && loops < MaxTransitions);
 
-    /* All done. Delete resources, and close down SDL_sound and OpenAL. */
+    /* All done. Delete resources, and close down OpenAL. */
     alDeleteSources(1, &source);
     alDeleteAuxiliaryEffectSlots(2, slots);
     alDeleteEffects(2, effects);
     alDeleteFilters(1, &direct_filter);
     alDeleteBuffers(1, &buffer);
 
-    Sound_Quit();
     CloseAL();
 
     return 0;
diff --git a/examples/alplay.c b/examples/alplay.c
index af1c313c..70314d47 100644
--- a/examples/alplay.c
+++ b/examples/alplay.c
@@ -158,7 +158,7 @@ int main(int argc, char **argv)
     } while(alGetError() == AL_NO_ERROR && state == AL_PLAYING);
     printf("\n");
 
-    /* All done. Delete resources, and close down SDL_sound and OpenAL. */
+    /* All done. Delete resources, and close down OpenAL. */
     alDeleteSources(1, &source);
     alDeleteBuffers(1, &buffer);
 
diff --git a/examples/alreverb.c b/examples/alreverb.c
index d789dffe..1f30a450 100644
--- a/examples/alreverb.c
+++ b/examples/alreverb.c
@@ -24,12 +24,13 @@
 
 /* This file contains an example for applying reverb to a sound. */
 
-#include <stdio.h>
 #include <assert.h>
+#include <inttypes.h>
+#include <limits.h>
+#include <stdio.h>
+#include <stdlib.h>
 
-#include "SDL_sound.h"
-#include "SDL_audio.h"
-#include "SDL_stdinc.h"
+#include "sndfile.h"
 
 #include "AL/al.h"
 #include "AL/alc.h"
@@ -149,68 +150,62 @@ static ALuint LoadEffect(const EFXEAXREVERBPROPERTIES *reverb)
  */
 static ALuint LoadSound(const char *filename)
 {
-    Sound_Sample *sample;
     ALenum err, format;
     ALuint buffer;
-    Uint32 slen;
-
-    /* Open the audio file */
-    sample = Sound_NewSampleFromFile(filename, NULL, 65536);
-    if(!sample)
+    SNDFILE *sndfile;
+    SF_INFO sfinfo;
+    short *membuf;
+    sf_count_t num_frames;
+    ALsizei num_bytes;
+
+    /* Open the audio file and check that it's usable. */
+    sndfile = sf_open(filename, SFM_READ, &sfinfo);
+    if(!sndfile)
     {
-        fprintf(stderr, "Could not open audio in %s\n", filename);
+        fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(sndfile));
         return 0;
     }
-
-    /* Get the sound format, and figure out the OpenAL format */
-    if(sample->actual.channels == 1)
-    {
-        if(sample->actual.format == AUDIO_U8)
-            format = AL_FORMAT_MONO8;
-        else if(sample->actual.format == AUDIO_S16SYS)
-            format = AL_FORMAT_MONO16;
-        else
-        {
-            fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format);
-            Sound_FreeSample(sample);
-            return 0;
-        }
-    }
-    else if(sample->actual.channels == 2)
+    if(sfinfo.frames < 1 || sfinfo.frames > (sf_count_t)(INT_MAX/sizeof(short))/sfinfo.channels)
     {
-        if(sample->actual.format == AUDIO_U8)
-            format = AL_FORMAT_STEREO8;
-        else if(sample->actual.format == AUDIO_S16SYS)
-            format = AL_FORMAT_STEREO16;
-        else
-        {
-            fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format);
-            Sound_FreeSample(sample);
-            return 0;
-        }
+        fprintf(stderr, "Bad sample count in %s (%" PRId64 ")\n", filename, sfinfo.frames);
+        sf_close(sndfile);
+        return 0;
     }
+
+    /* Get the sound format, and figure out the OpenAL format */
+    if(sfinfo.channels == 1)
+        format = AL_FORMAT_MONO16;
+    else if(sfinfo.channels == 2)
+        format = AL_FORMAT_STEREO16;
     else
     {
-        fprintf(stderr, "Unsupported channel count: %d\n", sample->actual.channels);
-        Sound_FreeSample(sample);
+        fprintf(stderr, "Unsupported channel count: %d\n", sfinfo.channels);
+        sf_close(sndfile);
         return 0;
     }
 
-    /* Decode the whole audio stream to a buffer. */
-    slen = Sound_DecodeAll(sample);
-    if(!sample->buffer || slen == 0)
+    /* Decode the whole audio file to a buffer. */
+    membuf = malloc((size_t)(sfinfo.frames * sfinfo.channels) * sizeof(short));
+
+    num_frames = sf_readf_short(sndfile, membuf, sfinfo.frames);
+    if(num_frames < 1)
     {
-        fprintf(stderr, "Failed to read audio from %s\n", filename);
-        Sound_FreeSample(sample);
+        free(membuf);
+        sf_close(sndfile);
+        fprintf(stderr, "Failed to read samples in %s (%" PRId64 ")\n", filename, num_frames);
         return 0;
     }
+    num_bytes = (ALsizei)(num_frames * sfinfo.channels) * (ALsizei)sizeof(short);
 
     /* Buffer the audio data into a new buffer object, then free the data and
-     * close the file. */
+     * close the file.
+     */
     buffer = 0;
     alGenBuffers(1, &buffer);
-    alBufferData(buffer, format, sample->buffer, (ALsizei)slen, (ALsizei)sample->actual.rate);
-    Sound_FreeSample(sample);
+    alBufferData(buffer, format, membuf, num_bytes, sfinfo.samplerate);
+
+    free(membuf);
+    sf_close(sndfile);
 
     /* Check if an error occured, and clean up if so. */
     err = alGetError();
@@ -278,15 +273,11 @@ int main(int argc, char **argv)
     LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTFV, alGetAuxiliaryEffectSlotfv);
 #undef LOAD_PROC
 
-    /* Initialize SDL_sound. */
-    Sound_Init();
-
     /* Load the sound into a buffer. */
     buffer = LoadSound(argv[0]);
     if(!buffer)
     {
         CloseAL();
-        Sound_Quit();
         return 1;
     }
 
@@ -295,7 +286,6 @@ int main(int argc, char **argv)
     if(!effect)
     {
         alDeleteBuffers(1, &buffer);
-        Sound_Quit();
         CloseAL();
         return 1;
     }
@@ -330,13 +320,12 @@ int main(int argc, char **argv)
         alGetSourcei(source, AL_SOURCE_STATE, &state);
     } while(alGetError() == AL_NO_ERROR && state == AL_PLAYING);
 
-    /* All done. Delete resources, and close down SDL_sound and OpenAL. */
+    /* All done. Delete resources, and close down OpenAL. */
     alDeleteSources(1, &source);
     alDeleteAuxiliaryEffectSlots(1, &slot);
     alDeleteEffects(1, &effect);
     alDeleteBuffers(1, &buffer);
 
-    Sound_Quit();
     CloseAL();
 
     return 0;
diff --git a/examples/alstream.c b/examples/alstream.c
index 56505ddb..69be236a 100644
--- a/examples/alstream.c
+++ b/examples/alstream.c
@@ -24,32 +24,24 @@
 
 /* This file contains a relatively simple streaming audio player. */
 
-#include <string.h>
-#include <stdlib.h>
-#include <stdio.h>
 #include <assert.h>
+#include <inttypes.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
 
-#include "SDL_sound.h"
-#include "SDL_audio.h"
-#include "SDL_stdinc.h"
+#include "sndfile.h"
 
 #include "AL/al.h"
 
 #include "common/alhelpers.h"
 
 
-#ifndef SDL_AUDIO_MASK_BITSIZE
-#define SDL_AUDIO_MASK_BITSIZE (0xFF)
-#endif
-#ifndef SDL_AUDIO_BITSIZE
-#define SDL_AUDIO_BITSIZE(x) (x & SDL_AUDIO_MASK_BITSIZE)
-#endif
-
 /* Define the number of buffers and buffer size (in milliseconds) to use. 4
- * buffers with 200ms each gives a nice per-chunk size, and lets the queue last
- * for almost one second. */
+ * buffers with 8192 samples each gives a nice per-chunk size, and lets the
+ * queue last for almost one second at 44.1khz. */
 #define NUM_BUFFERS 4
-#define BUFFER_TIME_MS 200
+#define BUFFER_SAMPLES 8192
 
 typedef struct StreamPlayer {
     /* These are the buffers and source to play out through OpenAL with */
@@ -57,11 +49,12 @@ typedef struct StreamPlayer {
     ALuint source;
 
     /* Handle for the audio file */
-    Sound_Sample *sample;
+    SNDFILE *sndfile;
+    SF_INFO sfinfo;
+    short *membuf;
 
-    /* The format of the output stream */
+    /* The format of the output stream (sample rate is in sfinfo) */
     ALenum format;
-    ALsizei srate;
 } StreamPlayer;
 
 static StreamPlayer *NewPlayer(void);
@@ -118,73 +111,46 @@ static void DeletePlayer(StreamPlayer *player)
  * it will be closed first. */
 static int OpenPlayerFile(StreamPlayer *player, const char *filename)
 {
-    Uint32 frame_size;
+    size_t frame_size;
 
     ClosePlayerFile(player);
 
-    /* Open the file and get the first stream from it */
-    player->sample = Sound_NewSampleFromFile(filename, NULL, 0);
-    if(!player->sample)
+    /* Open the audio file and check that it's usable. */
+    player->sndfile = sf_open(filename, SFM_READ, &player->sfinfo);
+    if(!player->sndfile)
     {
-        fprintf(stderr, "Could not open audio in %s\n", filename);
-        goto error;
+        fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(NULL));
+        return 0;
     }
 
-    /* Get the stream format, and figure out the OpenAL format */
-    if(player->sample->actual.channels == 1)
-    {
-        if(player->sample->actual.format == AUDIO_U8)
-            player->format = AL_FORMAT_MONO8;
-        else if(player->sample->actual.format == AUDIO_S16SYS)
-            player->format = AL_FORMAT_MONO16;
-        else
-        {
-            fprintf(stderr, "Unsupported sample format: 0x%04x\n", player->sample->actual.format);
-            goto error;
-        }
-    }
-    else if(player->sample->actual.channels == 2)
-    {
-        if(player->sample->actual.format == AUDIO_U8)
-            player->format = AL_FORMAT_STEREO8;
-        else if(player->sample->actual.format == AUDIO_S16SYS)
-            player->format = AL_FORMAT_STEREO16;
-        else
-        {
-            fprintf(stderr, "Unsupported sample format: 0x%04x\n", player->sample->actual.format);
-            goto error;
-        }
-    }
+    /* Get the sound format, and figure out the OpenAL format */
+    if(player->sfinfo.channels == 1)
+        player->format = AL_FORMAT_MONO16;
+    else if(player->sfinfo.channels == 2)
+        player->format = AL_FORMAT_STEREO16;
     else
     {
-        fprintf(stderr, "Unsupported channel count: %d\n", player->sample->actual.channels);
-        goto error;
+        fprintf(stderr, "Unsupported channel count: %d\n", player->sfinfo.channels);
+        sf_close(player->sndfile);
+        player->sndfile = NULL;
+        return 0;
     }
-    player->srate = (ALsizei)player->sample->actual.rate;
-
-    frame_size = player->sample->actual.channels *
-        SDL_AUDIO_BITSIZE(player->sample->actual.format) / 8;
 
-    /* Set the buffer size, given the desired millisecond length. */
-    Sound_SetBufferSize(player->sample, (Uint32)((Uint64)player->srate*BUFFER_TIME_MS/1000) *
-                                        frame_size);
+    frame_size = (size_t)(BUFFER_SAMPLES * player->sfinfo.channels) * sizeof(short);
+    player->membuf = malloc(frame_size);
 
     return 1;
-
-error:
-    if(player->sample)
-        Sound_FreeSample(player->sample);
-    player->sample = NULL;
-
-    return 0;
 }
 
 /* Closes the audio file stream */
 static void ClosePlayerFile(StreamPlayer *player)
 {
-    if(player->sample)
-        Sound_FreeSample(player->sample);
-    player->sample = NULL;
+    if(player->sndfile)
+        sf_close(player->sndfile);
+    player->sndfile = NULL;
+
+    free(player->membuf);
+    player->membuf = NULL;
 }
 
 
@@ -201,11 +167,12 @@ static int StartPlayer(StreamPlayer *player)
     for(i = 0;i < NUM_BUFFERS;i++)
     {
         /* Get some data to give it to the buffer */
-        Uint32 slen = Sound_Decode(player->sample);
-        if(slen == 0) break;
+        sf_count_t slen = sf_readf_short(player->sndfile, player->membuf, BUFFER_SAMPLES);
+        if(slen < 1) break;
 
-        alBufferData(player->buffers[i], player->format, player->sample->buffer, (ALsizei)slen,
-            player->srate);
+        slen *= player->sfinfo.channels * (sf_count_t)sizeof(short);
+        alBufferData(player->buffers[i], player->format, player->membuf, (ALsizei)slen,
+            player->sfinfo.samplerate);
     }
     if(alGetError() != AL_NO_ERROR)
     {
@@ -242,21 +209,19 @@ static int UpdatePlayer(StreamPlayer *player)
     while(processed > 0)
     {
         ALuint bufid;
-        Uint32 slen;
+        sf_count_t slen;
 
         alSourceUnqueueBuffers(player->source, 1, &bufid);
         processed--;
 
-        if((player->sample->flags&(SOUND_SAMPLEFLAG_EOF|SOUND_SAMPLEFLAG_ERROR)))
-            continue;
-
         /* Read the next chunk of data, refill the buffer, and queue it
          * back on the source */
-        slen = Sound_Decode(player->sample);
+        slen = sf_readf_short(player->sndfile, player->membuf, BUFFER_SAMPLES);
         if(slen > 0)
         {
-            alBufferData(bufid, player->format, player->sample->buffer, (ALsizei)slen,
-                player->srate);
+            slen *= player->sfinfo.channels * (sf_count_t)sizeof(short);
+            alBufferData(bufid, player->format, player->membuf, (ALsizei)slen,
+                player->sfinfo.samplerate);
             alSourceQueueBuffers(player->source, 1, &bufid);
         }
         if(alGetError() != AL_NO_ERROR)
@@ -304,8 +269,6 @@ int main(int argc, char **argv)
     if(InitAL(&argv, &argc) != 0)
         return 1;
 
-    Sound_Init();
-
     player = NewPlayer();
 
     /* Play each file listed on the command line */
@@ -323,7 +286,8 @@ int main(int argc, char **argv)
         else
             namepart = argv[i];
 
-        printf("Playing: %s (%s, %dhz)\n", namepart, FormatName(player->format), player->srate);
+        printf("Playing: %s (%s, %dhz)\n", namepart, FormatName(player->format),
+            player->sfinfo.samplerate);
         fflush(stdout);
 
         if(!StartPlayer(player))
@@ -340,11 +304,10 @@ int main(int argc, char **argv)
     }
     printf("Done.\n");
 
-    /* All files done. Delete the player, and close down SDL_sound and OpenAL */
+    /* All files done. Delete the player, and close down OpenAL */
     DeletePlayer(player);
     player = NULL;
 
-    Sound_Quit();
     CloseAL();
 
     return 0;
diff --git a/examples/alstreamcb.cpp b/examples/alstreamcb.cpp
index fbc3d02d..8fb1f102 100644
--- a/examples/alstreamcb.cpp
+++ b/examples/alstreamcb.cpp
@@ -36,9 +36,7 @@
 #include <thread>
 #include <vector>
 
-#include "SDL_sound.h"
-#include "SDL_audio.h"
-#include "SDL_stdinc.h"
+#include "sndfile.h"
 
 #include "AL/al.h"
 #include "AL/alc.h"
@@ -46,14 +44,6 @@
 #include "common/alhelpers.h"
 
 
-#ifndef SDL_AUDIO_MASK_BITSIZE
-#define SDL_AUDIO_MASK_BITSIZE (0xFF)
-#endif
-#ifndef SDL_AUDIO_BITSIZE
-#define SDL_AUDIO_BITSIZE(x) (x & SDL_AUDIO_MASK_BITSIZE)
-#endif
-
-
 #ifndef AL_SOFT_callback_buffer
 #define AL_SOFT_callback_buffer
 typedef unsigned int ALbitfieldSOFT;
@@ -87,13 +77,12 @@ struct StreamPlayer {
     size_t mStartOffset{0};
 
     /* Handle for the audio file to decode. */
-    Sound_Sample *mSample{nullptr};
-    Uint32 mAvailableData{0};
+    SNDFILE *mSndfile{nullptr};
+    SF_INFO mSfInfo{};
     size_t mDecoderOffset{0};
 
     /* The format of the callback samples. */
     ALenum mFormat;
-    ALsizei mSampleRate;
 
     StreamPlayer()
     {
@@ -111,18 +100,18 @@ struct StreamPlayer {
     {
         alDeleteSources(1, &mSource);
         alDeleteBuffers(1, &mBuffer);
-        if(mSample)
-            Sound_FreeSample(mSample);
+        if(mSndfile)
+            sf_close(mSndfile);
     }
 
     void close()
     {
-        if(mSample)
+        if(mSndfile)
         {
             alSourceRewind(mSource);
             alSourcei(mSource, AL_BUFFER, 0);
-            Sound_FreeSample(mSample);
-            mSample = nullptr;
+            sf_close(mSndfile);
+            mSndfile = nullptr;
         }
     }
 
@@ -130,50 +119,30 @@ struct StreamPlayer {
     {
         close();
 
-        /* Open the file in its normal format. */
-        mSample = Sound_NewSampleFromFile(filename, nullptr, 0);
-        if(!mSample)
+        /* Open the file and figure out the OpenAL format. */
+        mSndfile = sf_open(filename, SFM_READ, &mSfInfo);
+        if(!mSndfile)
         {
-            fprintf(stderr, "Could not open audio in %s\n", filename);
+            fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(mSndfile));
             return false;
         }
 
-        /* Figure out the OpenAL format from the sample's format. */
         mFormat = AL_NONE;
-        if(mSample->actual.channels == 1)
-        {
-            if(mSample->actual.format == AUDIO_U8)
-                mFormat = AL_FORMAT_MONO8;
-            else if(mSample->actual.format == AUDIO_S16SYS)
-                mFormat = AL_FORMAT_MONO16;
-        }
-        else if(mSample->actual.channels == 2)
-        {
-            if(mSample->actual.format == AUDIO_U8)
-                mFormat = AL_FORMAT_STEREO8;
-            else if(mSample->actual.format == AUDIO_S16SYS)
-                mFormat = AL_FORMAT_STEREO16;
-        }
-        if(!mFormat)
+        if(mSfInfo.channels == 1)
+            mFormat = AL_FORMAT_MONO16;
+        else if(mSfInfo.channels == 2)
+            mFormat = AL_FORMAT_STEREO16;
+        else
         {
-            fprintf(stderr, "Unsupported sample format: 0x%04x, %d channels\n",
-                mSample->actual.format, mSample->actual.channels);
-            Sound_FreeSample(mSample);
-            mSample = nullptr;
+            fprintf(stderr, "Unsupported channel count: %d\n", mSfInfo.channels);
+            sf_close(mSndfile);
+            mSndfile = nullptr;
 
             return false;
         }
-        mSampleRate = static_cast<ALsizei>(mSample->actual.rate);
-
-        const auto frame_size = Uint32{mSample->actual.channels} *
-            SDL_AUDIO_BITSIZE(mSample->actual.format) / 8;
-
-        /* Set a 50ms decode buffer size. */
-        Sound_SetBufferSize(mSample, static_cast<Uint32>(mSampleRate)*50/1000 * frame_size);
-        mAvailableData = 0;
 
         /* Set a 1s ring buffer size. */
-        mBufferDataSize = static_cast<Uint32>(mSampleRate) * size_t{frame_size};
+        mBufferDataSize = static_cast<ALuint>(mSfInfo.samplerate*mSfInfo.channels) * sizeof(short);
         mBufferData.reset(new ALbyte[mBufferDataSize]);
         mReadPos.store(0, std::memory_order_relaxed);
         mWritePos.store(0, std::memory_order_relaxed);
@@ -239,34 +208,27 @@ struct StreamPlayer {
 
     bool prepare()
     {
-        alBufferCallbackSOFT(mBuffer, mFormat, mSampleRate, bufferCallbackC, this, 0);
+        alBufferCallbackSOFT(mBuffer, mFormat, mSfInfo.samplerate, bufferCallbackC, this, 0);
         alSourcei(mSource, AL_BUFFER, static_cast<ALint>(mBuffer));
         if(ALenum err{alGetError()})
         {
             fprintf(stderr, "Failed to set callback: %s (0x%04x)\n", alGetString(err), err);
             return false;
         }
-
-        mAvailableData = Sound_Decode(mSample);
-        if(!mAvailableData)
-            fprintf(stderr, "Failed to decode any samples: %s\n", Sound_GetError());
-        return mAvailableData != 0;
+        return true;
     }
 
     bool update()
     {
-        constexpr int BadFlags{SOUND_SAMPLEFLAG_EOF | SOUND_SAMPLEFLAG_ERROR};
-
         ALenum state;
         ALint pos;
         alGetSourcei(mSource, AL_SAMPLE_OFFSET, &pos);
         alGetSourcei(mSource, AL_SOURCE_STATE, &state);
 
+        const size_t frame_size{static_cast<ALuint>(mSfInfo.channels) * sizeof(short)};
         size_t woffset{mWritePos.load(std::memory_order_acquire)};
         if(state != AL_INITIAL)
         {
-            const auto frame_size = Uint32{mSample->actual.channels} *
-                SDL_AUDIO_BITSIZE(mSample->actual.format) / 8;
             const size_t roffset{mReadPos.load(std::memory_order_relaxed)};
             const size_t readable{((woffset >= roffset) ? woffset : (mBufferDataSize+woffset)) -
                 roffset};
@@ -276,15 +238,17 @@ struct StreamPlayer {
              * the playback offset the source was started with.
              */
             const size_t curtime{((state==AL_STOPPED) ? (mDecoderOffset-readable) / frame_size
-                : (static_cast<ALuint>(pos) + mStartOffset/frame_size)) / mSample->actual.rate};
+                : (static_cast<ALuint>(pos) + mStartOffset/frame_size))
+                / static_cast<ALuint>(mSfInfo.samplerate)};
             printf("\r%3zus (%3zu%% full)", curtime, readable * 100 / mBufferDataSize);
         }
         else
             fputs("Starting...", stdout);
         fflush(stdout);
 
-        while(mAvailableData > 0)
+        while(!sf_error(mSndfile))
         {
+            size_t read_bytes;
             const size_t roffset{mReadPos.load(std::memory_order_relaxed)};
             if(roffset > woffset)
             {
@@ -294,45 +258,39 @@ struct StreamPlayer {
                  * instead of full.
                  */
                 const size_t writable{roffset-woffset-1};
-                /* Don't copy the sample data if it can't all fit. */
-                if(writable < mAvailableData) break;
+                if(writable < frame_size) break;
+
+                sf_count_t num_frames{sf_readf_short(mSndfile,
+                    reinterpret_cast<short*>(&mBufferData[woffset]),
+                    static_cast<sf_count_t>(writable/frame_size))};
+                if(num_frames < 1) break;
 
-                memcpy(&mBufferData[woffset], mSample->buffer, mAvailableData);
-                woffset += mAvailableData;
+                read_bytes = static_cast<size_t>(num_frames) * frame_size;
+                woffset += read_bytes;
             }
             else
             {
                 /* If the read offset is at or behind the write offset, the
                  * writeable area (might) wrap around. Make sure the sample
-                 * data can fit, and calculate how much goes in front and in
-                 * back.
+                 * data can fit, and calculate how much can go in front before
+                 * wrapping.
                  */
-                const size_t writable{mBufferDataSize+roffset-woffset-1};
-                if(writable < mAvailableData) break;
+                const size_t writable{!roffset ? mBufferDataSize-woffset-1 :
+                    (mBufferDataSize-woffset)};
+                if(writable < frame_size) break;
 
-                const size_t todo1{std::min<size_t>(mAvailableData, mBufferDataSize-woffset)};
-                const size_t todo2{mAvailableData - todo1};
+                sf_count_t num_frames{sf_readf_short(mSndfile,
+                    reinterpret_cast<short*>(&mBufferData[woffset]),
+                    static_cast<sf_count_t>(writable/frame_size))};
+                if(num_frames < 1) break;
 
-                memcpy(&mBufferData[woffset], mSample->buffer, todo1);
-                woffset += todo1;
+                read_bytes = static_cast<size_t>(num_frames) * frame_size;
+                woffset += read_bytes;
                 if(woffset == mBufferDataSize)
-                {
                     woffset = 0;
-                    if(todo2 > 0)
-                    {
-                        memcpy(&mBufferData[woffset], static_cast<ALbyte*>(mSample->buffer)+todo1,
-                            todo2);
-                        woffset += todo2;
-                    }
-                }
             }
             mWritePos.store(woffset, std::memory_order_release);
-            mDecoderOffset += mAvailableData;
-
-            if(!(mSample->flags&BadFlags))
-                mAvailableData = Sound_Decode(mSample);
-            else
-                mAvailableData = 0;
+            mDecoderOffset += read_bytes;
         }
 
         if(state != AL_PLAYING && state != AL_PAUSED)
@@ -364,15 +322,14 @@ struct StreamPlayer {
 
 int main(int argc, char **argv)
 {
-    /* A simple RAII container for OpenAL and SDL_sound startup and shutdown. */
+    /* A simple RAII container for OpenAL startup and shutdown. */
     struct AudioManager {
         AudioManager(char ***argv_, int *argc_)
         {
             if(InitAL(argv_, argc_) != 0)
                 throw std::runtime_error{"Failed to initialize OpenAL"};
-            Sound_Init();
         }
-        ~AudioManager() { Sound_Quit(); CloseAL(); }
+        ~AudioManager() { CloseAL(); }
     };
 
     /* Print out usage if no arguments were specified */
@@ -413,7 +370,7 @@ int main(int argc, char **argv)
             namepart = argv[i];
 
         printf("Playing: %s (%s, %dhz)\n", namepart, FormatName(player->mFormat),
-            player->mSampleRate);
+            player->mSfInfo.samplerate);
         fflush(stdout);
 
         if(!player->prepare())