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Convert the examples from SDL_sound to libsndfile

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This commit is contained in:
Chris Robinson 2020-03-24 15:46:47 -07:00
parent 586bc94d51
commit cae78e79e8
9 changed files with 307 additions and 876 deletions
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83
CMakeLists.txt
View File

@ -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()
IF(SDL2_FOUND)
FIND_PACKAGE(SDL_sound)
FIND_PACKAGE(SndFile)
FIND_PACKAGE(SDL2)
IF(SDL2_FOUND)
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})
@ -1495,59 +1518,19 @@ 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_INCLUDE_DIRECTORIES(alloopback PRIVATE ${SDL2_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})
PRIVATE ${LINKER_FLAGS} ${SDL2_LIBRARY} ex-common ${MATH_LIB})
IF(ALSOFT_INSTALL)
INSTALL(TARGETS alstream alreverb almultireverb allatency alloopback alhrtf
INSTALL(TARGETS alloopback
RUNTIME DESTINATION ${CMAKE_INSTALL_BINDIR}
LIBRARY DESTINATION ${CMAKE_INSTALL_LIBDIR}
ARCHIVE DESTINATION ${CMAKE_INSTALL_LIBDIR})
ENDIF()
MESSAGE(STATUS "Building SDL_sound example programs")
ENDIF()
MESSAGE(STATUS "Building SDL example programs")
SET(FFVER_OK FALSE)
IF(FFMPEG_FOUND)

429
cmake/FindSDL_sound.cmake
View File

@ -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)

89
examples/alhrtf.c
View File

@ -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;
SNDFILE *sndfile;
SF_INFO sfinfo;
short *membuf;
sf_count_t num_frames;
ALsizei num_bytes;
/* Open the audio file */
sample = Sound_NewSampleFromFile(filename, NULL, 65536);
if(!sample)
/* 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;
}
if(sfinfo.frames < 1 || sfinfo.frames > (sf_count_t)(INT_MAX/sizeof(short))/sfinfo.channels)
{
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(sample->actual.channels == 1)
{
if(sample->actual.format == AUDIO_U8)
format = AL_FORMAT_MONO8;
else if(sample->actual.format == AUDIO_S16SYS)
if(sfinfo.channels == 1)
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(sample->actual.format == AUDIO_U8)
format = AL_FORMAT_STEREO8;
else if(sample->actual.format == AUDIO_S16SYS)
else if(sfinfo.channels == 2)
format = AL_FORMAT_STEREO16;
else
{
fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format);
Sound_FreeSample(sample);
return 0;
}
}
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;

91
examples/allatency.c
View File

@ -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;
SNDFILE *sndfile;
SF_INFO sfinfo;
short *membuf;
sf_count_t num_frames;
ALsizei num_bytes;
/* Open the audio file */
sample = Sound_NewSampleFromFile(filename, NULL, 65536);
if(!sample)
/* 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;
}
if(sfinfo.frames < 1 || sfinfo.frames > (sf_count_t)(INT_MAX/sizeof(short))/sfinfo.channels)
{
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(sample->actual.channels == 1)
{
if(sample->actual.format == AUDIO_U8)
format = AL_FORMAT_MONO8;
else if(sample->actual.format == AUDIO_S16SYS)
if(sfinfo.channels == 1)
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(sample->actual.format == AUDIO_U8)
format = AL_FORMAT_STEREO8;
else if(sample->actual.format == AUDIO_S16SYS)
else if(sfinfo.channels == 2)
format = AL_FORMAT_STEREO16;
else
{
fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format);
Sound_FreeSample(sample);
return 0;
}
}
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;

90
examples/almultireverb.c
View File

@ -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;
SNDFILE *sndfile;
SF_INFO sfinfo;
short *membuf;
sf_count_t num_frames;
ALsizei num_bytes;
/* Open the audio file */
sample = Sound_NewSampleFromFile(filename, NULL, 65536);
if(!sample)
/* 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;
}
if(sfinfo.frames < 1 || sfinfo.frames > (sf_count_t)(INT_MAX/sizeof(short))/sfinfo.channels)
{
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(sample->actual.channels == 1)
{
if(sample->actual.format == AUDIO_U8)
format = AL_FORMAT_MONO8;
else if(sample->actual.format == AUDIO_S16SYS)
if(sfinfo.channels == 1)
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(sample->actual.format == AUDIO_U8)
format = AL_FORMAT_STEREO8;
else if(sample->actual.format == AUDIO_S16SYS)
else if(sfinfo.channels == 2)
format = AL_FORMAT_STEREO16;
else
{
fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format);
Sound_FreeSample(sample);
return 0;
}
}
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;

2
examples/alplay.c
View File

@ -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);

91
examples/alreverb.c
View File

@ -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;
SNDFILE *sndfile;
SF_INFO sfinfo;
short *membuf;
sf_count_t num_frames;
ALsizei num_bytes;
/* Open the audio file */
sample = Sound_NewSampleFromFile(filename, NULL, 65536);
if(!sample)
/* 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;
}
if(sfinfo.frames < 1 || sfinfo.frames > (sf_count_t)(INT_MAX/sizeof(short))/sfinfo.channels)
{
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(sample->actual.channels == 1)
{
if(sample->actual.format == AUDIO_U8)
format = AL_FORMAT_MONO8;
else if(sample->actual.format == AUDIO_S16SYS)
if(sfinfo.channels == 1)
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(sample->actual.format == AUDIO_U8)
format = AL_FORMAT_STEREO8;
else if(sample->actual.format == AUDIO_S16SYS)
else if(sfinfo.channels == 2)
format = AL_FORMAT_STEREO16;
else
{
fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format);
Sound_FreeSample(sample);
return 0;
}
}
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;

129
examples/alstream.c
View File

@ -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)
/* Get the sound format, and figure out the OpenAL format */
if(player->sfinfo.channels == 1)
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)
else if(player->sfinfo.channels == 2)
player->format = AL_FORMAT_STEREO16;
else
{
fprintf(stderr, "Unsupported sample format: 0x%04x\n", player->sample->actual.format);
goto error;
fprintf(stderr, "Unsupported channel count: %d\n", player->sfinfo.channels);
sf_close(player->sndfile);
player->sndfile = NULL;
return 0;
}
}
else
{
fprintf(stderr, "Unsupported channel count: %d\n", player->sample->actual.channels);
goto error;
}
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;

141
examples/alstreamcb.cpp
View File

@ -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)
if(mSfInfo.channels == 1)
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)
else if(mSfInfo.channels == 2)
mFormat = AL_FORMAT_STEREO16;
}
if(!mFormat)
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;
memcpy(&mBufferData[woffset], mSample->buffer, mAvailableData);
woffset += mAvailableData;
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;
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())