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Use SDL_sound when available to decode files
This commit is contained in:
parent
99209c4a2a
commit
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6 changed files with 2744 additions and 1578 deletions
382
FindSDL_sound.cmake
Normal file
382
FindSDL_sound.cmake
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@ -0,0 +1,382 @@
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# - Locates the SDL_sound library
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#
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# This module depends on SDL being found and
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# must be called AFTER FindSDL.cmake is called.
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#
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# This module defines
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# SDL_SOUND_INCLUDE_DIR, where to find SDL_sound.h
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# SDL_SOUND_FOUND, if false, do not try to link to SDL_sound
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# SDL_SOUND_LIBRARIES, this contains the list of libraries that you need
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# to link against. This is a read-only variable and is marked INTERNAL.
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# SDL_SOUND_EXTRAS, this is an optional variable for you to add your own
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# flags to SDL_SOUND_LIBRARIES. This is prepended to SDL_SOUND_LIBRARIES.
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# This is available mostly for cases this module failed to anticipate for
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# and you must add additional flags. This is marked as ADVANCED.
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# SDL_SOUND_VERSION_STRING, human-readable string containing the version of SDL_sound
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#
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# This module also defines (but you shouldn't need to use directly)
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# SDL_SOUND_LIBRARY, the name of just the SDL_sound library you would link
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# against. Use SDL_SOUND_LIBRARIES for you link instructions and not this one.
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# And might define the following as needed
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# MIKMOD_LIBRARY
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# MODPLUG_LIBRARY
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# OGG_LIBRARY
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# VORBIS_LIBRARY
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# SMPEG_LIBRARY
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# FLAC_LIBRARY
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# SPEEX_LIBRARY
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#
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# Typically, you should not use these variables directly, and you should use
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# SDL_SOUND_LIBRARIES which contains SDL_SOUND_LIBRARY and the other audio libraries
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# (if needed) to successfully compile on your system.
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#
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# Created by Eric Wing.
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# This module is a bit more complicated than the other FindSDL* family modules.
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# The reason is that SDL_sound can be compiled in a large variety of different ways
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# which are independent of platform. SDL_sound may dynamically link against other 3rd
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# party libraries to get additional codec support, such as Ogg Vorbis, SMPEG, ModPlug,
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# MikMod, FLAC, Speex, and potentially others.
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# Under some circumstances which I don't fully understand,
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# there seems to be a requirement
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# that dependent libraries of libraries you use must also be explicitly
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# linked against in order to successfully compile. SDL_sound does not currently
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# have any system in place to know how it was compiled.
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# So this CMake module does the hard work in trying to discover which 3rd party
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# libraries are required for building (if any).
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# This module uses a brute force approach to create a test program that uses SDL_sound,
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# and then tries to build it. If the build fails, it parses the error output for
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# known symbol names to figure out which libraries are needed.
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#
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# Responds to the $SDLDIR and $SDLSOUNDDIR environmental variable that would
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# correspond to the ./configure --prefix=$SDLDIR used in building SDL.
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#
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# On OSX, this will prefer the Framework version (if found) over others.
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# People will have to manually change the cache values of
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# SDL_LIBRARY to override this selectionor set the CMake environment
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# CMAKE_INCLUDE_PATH to modify the search paths.
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#=============================================================================
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# Copyright 2005-2009 Kitware, Inc.
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# Copyright 2012 Benjamin Eikel
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#
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# Distributed under the OSI-approved BSD License (the "License");
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# see accompanying file Copyright.txt for details.
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#
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# This software is distributed WITHOUT ANY WARRANTY; without even the
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# implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
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# See the License for more information.
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#=============================================================================
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# (To distribute this file outside of CMake, substitute the full
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# License text for the above reference.)
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set(SDL_SOUND_EXTRAS "" CACHE STRING "SDL_sound extra flags")
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mark_as_advanced(SDL_SOUND_EXTRAS)
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# Find SDL_sound.h
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find_path(SDL_SOUND_INCLUDE_DIR SDL_sound.h
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HINTS
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ENV SDLSOUNDDIR
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ENV SDLDIR
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PATH_SUFFIXES SDL SDL12 SDL11
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)
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find_library(SDL_SOUND_LIBRARY
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NAMES SDL_sound
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HINTS
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ENV SDLSOUNDDIR
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ENV SDLDIR
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)
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if(SDL_FOUND AND SDL_SOUND_INCLUDE_DIR AND SDL_SOUND_LIBRARY)
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# CMake is giving me problems using TRY_COMPILE with the CMAKE_FLAGS
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# for the :STRING syntax if I have multiple values contained in a
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# single variable. This is a problem for the SDL_LIBRARY variable
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# because it does just that. When I feed this variable to the command,
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# only the first value gets the appropriate modifier (e.g. -I) and
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# the rest get dropped.
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# To get multiple single variables to work, I must separate them with a "\;"
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# I could go back and modify the FindSDL.cmake module, but that's kind of painful.
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# The solution would be to try something like:
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# set(SDL_TRY_COMPILE_LIBRARY_LIST "${SDL_TRY_COMPILE_LIBRARY_LIST}\;${CMAKE_THREAD_LIBS_INIT}")
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# Instead, it was suggested on the mailing list to write a temporary CMakeLists.txt
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# with a temporary test project and invoke that with TRY_COMPILE.
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# See message thread "Figuring out dependencies for a library in order to build"
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# 2005-07-16
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# try_compile(
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# MY_RESULT
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# ${CMAKE_BINARY_DIR}
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# ${PROJECT_SOURCE_DIR}/DetermineSoundLibs.c
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# CMAKE_FLAGS
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# -DINCLUDE_DIRECTORIES:STRING=${SDL_INCLUDE_DIR}\;${SDL_SOUND_INCLUDE_DIR}
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# -DLINK_LIBRARIES:STRING=${SDL_SOUND_LIBRARY}\;${SDL_LIBRARY}
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# OUTPUT_VARIABLE MY_OUTPUT
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# )
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# To minimize external dependencies, create a sdlsound test program
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# which will be used to figure out if additional link dependencies are
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# required for the link phase.
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file(WRITE ${PROJECT_BINARY_DIR}/CMakeTmp/DetermineSoundLibs.c
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"#include \"SDL_sound.h\"
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#include \"SDL.h\"
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int main(int argc, char* argv[])
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{
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Sound_AudioInfo desired;
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Sound_Sample* sample;
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SDL_Init(0);
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Sound_Init();
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/* This doesn't actually have to work, but Init() is a no-op
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* for some of the decoders, so this should force more symbols
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* to be pulled in.
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*/
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sample = Sound_NewSampleFromFile(argv[1], &desired, 4096);
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Sound_Quit();
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SDL_Quit();
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return 0;
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}"
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)
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# Calling
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# target_link_libraries(DetermineSoundLibs "${SDL_SOUND_LIBRARY} ${SDL_LIBRARY})
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# causes problems when SDL_LIBRARY looks like
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# /Library/Frameworks/SDL.framework;-framework Cocoa
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# The ;-framework Cocoa seems to be confusing CMake once the OS X
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# framework support was added. I was told that breaking up the list
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# would fix the problem.
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set(TMP_TRY_LIBS)
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foreach(lib ${SDL_SOUND_LIBRARY} ${SDL_LIBRARY})
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set(TMP_TRY_LIBS "${TMP_TRY_LIBS} \"${lib}\"")
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endforeach()
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# message("TMP_TRY_LIBS ${TMP_TRY_LIBS}")
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# Write the CMakeLists.txt and test project
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# Weird, this is still sketchy. If I don't quote the variables
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# in the TARGET_LINK_LIBRARIES, I seem to loose everything
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# in the SDL_LIBRARY string after the "-framework".
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# But if I quote the stuff in INCLUDE_DIRECTORIES, it doesn't work.
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file(WRITE ${PROJECT_BINARY_DIR}/CMakeTmp/CMakeLists.txt
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"cmake_minimum_required(VERSION 2.8)
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project(DetermineSoundLibs C)
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include_directories(${SDL_INCLUDE_DIR} ${SDL_SOUND_INCLUDE_DIR})
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add_executable(DetermineSoundLibs DetermineSoundLibs.c)
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target_link_libraries(DetermineSoundLibs ${TMP_TRY_LIBS})"
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)
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try_compile(
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MY_RESULT
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${PROJECT_BINARY_DIR}/CMakeTmp
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${PROJECT_BINARY_DIR}/CMakeTmp
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DetermineSoundLibs
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OUTPUT_VARIABLE MY_OUTPUT
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)
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# message("${MY_RESULT}")
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# message(${MY_OUTPUT})
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if(NOT MY_RESULT)
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# I expect that MPGLIB, VOC, WAV, AIFF, and SHN are compiled in statically.
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# I think Timidity is also compiled in statically.
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# I've never had to explcitly link against Quicktime, so I'll skip that for now.
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set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARY})
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# Find MikMod
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if("${MY_OUTPUT}" MATCHES "MikMod_")
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find_library(MIKMOD_LIBRARY
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NAMES libmikmod-coreaudio mikmod
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PATHS
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ENV MIKMODDIR
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ENV SDLSOUNDDIR
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ENV SDLDIR
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/sw
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/opt/local
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/opt/csw
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/opt
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PATH_SUFFIXES
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lib
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)
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if(MIKMOD_LIBRARY)
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set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARIES_TMP} ${MIKMOD_LIBRARY})
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endif(MIKMOD_LIBRARY)
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endif("${MY_OUTPUT}" MATCHES "MikMod_")
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# Find ModPlug
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if("${MY_OUTPUT}" MATCHES "MODPLUG_")
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find_library(MODPLUG_LIBRARY
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NAMES modplug
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PATHS
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ENV MODPLUGDIR
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ENV SDLSOUNDDIR
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ENV SDLDIR
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/sw
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/opt/local
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/opt/csw
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/opt
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PATH_SUFFIXES
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lib
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)
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if(MODPLUG_LIBRARY)
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set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARIES_TMP} ${MODPLUG_LIBRARY})
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endif()
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endif()
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# Find Ogg and Vorbis
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if("${MY_OUTPUT}" MATCHES "ov_")
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find_library(VORBIS_LIBRARY
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NAMES vorbis Vorbis VORBIS
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PATHS
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ENV VORBISDIR
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ENV OGGDIR
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ENV SDLSOUNDDIR
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ENV SDLDIR
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/sw
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/opt/local
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/opt/csw
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/opt
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PATH_SUFFIXES
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lib
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)
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if(VORBIS_LIBRARY)
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set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARIES_TMP} ${VORBIS_LIBRARY})
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endif()
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find_library(OGG_LIBRARY
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NAMES ogg Ogg OGG
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PATHS
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ENV OGGDIR
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ENV VORBISDIR
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ENV SDLSOUNDDIR
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ENV SDLDIR
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/sw
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/opt/local
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/opt/csw
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/opt
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PATH_SUFFIXES
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lib
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)
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if(OGG_LIBRARY)
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set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARIES_TMP} ${OGG_LIBRARY})
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endif()
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endif()
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# Find SMPEG
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if("${MY_OUTPUT}" MATCHES "SMPEG_")
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find_library(SMPEG_LIBRARY
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NAMES smpeg SMPEG Smpeg SMpeg
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PATHS
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ENV SMPEGDIR
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ENV SDLSOUNDDIR
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ENV SDLDIR
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/sw
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/opt/local
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/opt/csw
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/opt
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PATH_SUFFIXES
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lib
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)
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if(SMPEG_LIBRARY)
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set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARIES_TMP} ${SMPEG_LIBRARY})
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endif()
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endif()
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# Find FLAC
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if("${MY_OUTPUT}" MATCHES "FLAC_")
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find_library(FLAC_LIBRARY
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NAMES flac FLAC
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PATHS
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ENV FLACDIR
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ENV SDLSOUNDDIR
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ENV SDLDIR
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/sw
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/opt/local
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/opt/csw
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/opt
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PATH_SUFFIXES
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lib
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)
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if(FLAC_LIBRARY)
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set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARIES_TMP} ${FLAC_LIBRARY})
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endif()
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endif()
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# Hmmm...Speex seems to depend on Ogg. This might be a problem if
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# the TRY_COMPILE attempt gets blocked at SPEEX before it can pull
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# in the Ogg symbols. I'm not sure if I should duplicate the ogg stuff
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# above for here or if two ogg entries will screw up things.
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if("${MY_OUTPUT}" MATCHES "speex_")
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find_library(SPEEX_LIBRARY
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NAMES speex SPEEX
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PATHS
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ENV SPEEXDIR
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ENV SDLSOUNDDIR
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ENV SDLDIR
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/sw
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/opt/local
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/opt/csw
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/opt
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PATH_SUFFIXES
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lib
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)
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if(SPEEX_LIBRARY)
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set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARIES_TMP} ${SPEEX_LIBRARY})
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endif()
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# Find OGG (needed for Speex)
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# We might have already found Ogg for Vorbis, so skip it if so.
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if(NOT OGG_LIBRARY)
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find_library(OGG_LIBRARY
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NAMES ogg Ogg OGG
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PATHS
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ENV OGGDIR
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ENV VORBISDIR
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ENV SPEEXDIR
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ENV SDLSOUNDDIR
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ENV SDLDIR
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/sw
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/opt/local
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/opt/csw
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/opt
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PATH_SUFFIXES lib
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)
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if(OGG_LIBRARY)
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set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARIES_TMP} ${OGG_LIBRARY})
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endif()
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endif()
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endif()
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set(SDL_SOUND_LIBRARIES ${SDL_SOUND_EXTRAS} ${SDL_SOUND_LIBRARIES_TMP} CACHE INTERNAL "SDL_sound and dependent libraries")
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else()
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set(SDL_SOUND_LIBRARIES ${SDL_SOUND_EXTRAS} ${SDL_SOUND_LIBRARY} CACHE INTERNAL "SDL_sound and dependent libraries")
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endif()
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endif()
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if(SDL_SOUND_INCLUDE_DIR AND EXISTS "${SDL_SOUND_INCLUDE_DIR}/SDL_sound.h")
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file(STRINGS "${SDL_SOUND_INCLUDE_DIR}/SDL_sound.h" SDL_SOUND_VERSION_MAJOR_LINE REGEX "^#define[ \t]+SOUND_VER_MAJOR[ \t]+[0-9]+$")
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file(STRINGS "${SDL_SOUND_INCLUDE_DIR}/SDL_sound.h" SDL_SOUND_VERSION_MINOR_LINE REGEX "^#define[ \t]+SOUND_VER_MINOR[ \t]+[0-9]+$")
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file(STRINGS "${SDL_SOUND_INCLUDE_DIR}/SDL_sound.h" SDL_SOUND_VERSION_PATCH_LINE REGEX "^#define[ \t]+SOUND_VER_PATCH[ \t]+[0-9]+$")
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string(REGEX REPLACE "^#define[ \t]+SOUND_VER_MAJOR[ \t]+([0-9]+)$" "\\1" SDL_SOUND_VERSION_MAJOR "${SDL_SOUND_VERSION_MAJOR_LINE}")
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string(REGEX REPLACE "^#define[ \t]+SOUND_VER_MINOR[ \t]+([0-9]+)$" "\\1" SDL_SOUND_VERSION_MINOR "${SDL_SOUND_VERSION_MINOR_LINE}")
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string(REGEX REPLACE "^#define[ \t]+SOUND_VER_PATCH[ \t]+([0-9]+)$" "\\1" SDL_SOUND_VERSION_PATCH "${SDL_SOUND_VERSION_PATCH_LINE}")
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set(SDL_SOUND_VERSION_STRING ${SDL_SOUND_VERSION_MAJOR}.${SDL_SOUND_VERSION_MINOR}.${SDL_SOUND_VERSION_PATCH})
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unset(SDL_SOUND_VERSION_MAJOR_LINE)
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unset(SDL_SOUND_VERSION_MINOR_LINE)
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unset(SDL_SOUND_VERSION_PATCH_LINE)
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unset(SDL_SOUND_VERSION_MAJOR)
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unset(SDL_SOUND_VERSION_MINOR)
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unset(SDL_SOUND_VERSION_PATCH)
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endif()
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include(FindPackageHandleStandardArgs)
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FIND_PACKAGE_HANDLE_STANDARD_ARGS(SDL_sound
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REQUIRED_VARS SDL_SOUND_LIBRARY SDL_SOUND_INCLUDE_DIR
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VERSION_VAR SDL_SOUND_VERSION_STRING)
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@ -202,11 +202,22 @@ else( WIN32 )
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endif( WIN32 )
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set( OAL_SOURCES sound/oalsound.cpp )
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if( NOT NO_OPENAL )
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find_package( OpenAL )
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if( OPENAL_FOUND )
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include_directories( ${OPENAL_INCLUDE_DIR} )
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set( ZDOOM_LIBS ${OPENAL_LIBRARY} ${ZDOOM_LIBS} )
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find_package( SDL )
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if( SDL_FOUND )
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find_package( SDL_sound )
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if( SDL_SOUND_FOUND )
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set_source_files_properties( sound/oalsound.cpp PROPERTIES COMPILE_FLAGS "-DWITH_SDL_SOUND=1" )
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set( OAL_SOURCES ${OAL_SOURCES} sound/oalsdlsound.cpp )
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include_directories( ${SDL_SOUND_INCLUDE_DIR} )
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set( ZDOOM_LIBS ${SDL_SOUND_LIBRARIES} ${ZDOOM_LIBS} )
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endif( SDL_SOUND_FOUND )
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endif( SDL_FOUND )
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else( OPENAL_FOUND )
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set( NO_OPENAL ON )
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endif( OPENAL_FOUND )
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|
@ -900,7 +911,7 @@ add_executable( zdoom WIN32
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sound/music_softsynth_mididevice.cpp
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sound/music_timidity_mididevice.cpp
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sound/music_win_mididevice.cpp
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sound/oalsound.cpp
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${OAL_SOURCES}
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sound/music_pseudo_mididevice.cpp
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textures/animations.cpp
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textures/anim_switches.cpp
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|
|
494
src/sound/oalsdlsound.cpp
Normal file
494
src/sound/oalsdlsound.cpp
Normal file
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@ -0,0 +1,494 @@
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/*
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** oalsdlsound.cpp
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** Interface for SDL_sound; uses OpenAL
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**
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**---------------------------------------------------------------------------
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** Copyright 2008-2010 Chris Robinson
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||||
** All rights reserved.
|
||||
**
|
||||
** Redistribution and use in source and binary forms, with or without
|
||||
** modification, are permitted provided that the following conditions
|
||||
** are met:
|
||||
**
|
||||
** 1. Redistributions of source code must retain the above copyright
|
||||
** notice, this list of conditions and the following disclaimer.
|
||||
** 2. Redistributions in binary form must reproduce the above copyright
|
||||
** notice, this list of conditions and the following disclaimer in the
|
||||
** documentation and/or other materials provided with the distribution.
|
||||
** 3. The name of the author may not be used to endorse or promote products
|
||||
** derived from this software without specific prior written permission.
|
||||
**
|
||||
** THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
|
||||
** IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
|
||||
** OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED.
|
||||
** IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
|
||||
** INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT
|
||||
** NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
|
||||
** DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
|
||||
** THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
|
||||
** (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
|
||||
** THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
|
||||
**---------------------------------------------------------------------------
|
||||
**
|
||||
*/
|
||||
|
||||
#ifdef _WIN32
|
||||
#define WIN32_LEAN_AND_MEAN
|
||||
#include <windows.h>
|
||||
#define USE_WINDOWS_DWORD
|
||||
#endif
|
||||
|
||||
#include "doomstat.h"
|
||||
#include "templates.h"
|
||||
#include "oalsound.h"
|
||||
#include "oalsdlsound.h"
|
||||
#include "c_cvars.h"
|
||||
#include "c_dispatch.h"
|
||||
#include "i_system.h"
|
||||
#include "v_text.h"
|
||||
#include "gi.h"
|
||||
#include "actor.h"
|
||||
#include "r_state.h"
|
||||
#include "w_wad.h"
|
||||
#include "i_music.h"
|
||||
#include "i_musicinterns.h"
|
||||
|
||||
#include <algorithm>
|
||||
#include <memory>
|
||||
|
||||
|
||||
struct RWSubFile {
|
||||
private:
|
||||
FILE *fp;
|
||||
size_t start;
|
||||
size_t length;
|
||||
|
||||
public:
|
||||
RWSubFile(FILE *f, size_t offset, size_t len)
|
||||
: fp(f), start(offset), length(len)
|
||||
{
|
||||
fseek(fp, start, SEEK_SET);
|
||||
}
|
||||
~RWSubFile()
|
||||
{ fclose(fp); }
|
||||
|
||||
static int seek(SDL_RWops *context, int offset, int whence)
|
||||
{
|
||||
RWSubFile *self = static_cast<RWSubFile*>(context->hidden.unknown.data1);
|
||||
|
||||
if(whence == SEEK_END)
|
||||
{
|
||||
if(offset <= 0)
|
||||
offset = self->length + offset;
|
||||
}
|
||||
else if(whence == SEEK_CUR)
|
||||
offset = offset + ftell(self->fp) - self->start;
|
||||
else if(whence != SEEK_SET)
|
||||
{
|
||||
SDL_SetError("Invalid seek mode");
|
||||
return -1;
|
||||
}
|
||||
|
||||
if(offset >= 0 && size_t(offset) <= self->length)
|
||||
{
|
||||
if(fseek(self->fp, offset + self->start, SEEK_SET) == 0)
|
||||
return offset;
|
||||
}
|
||||
|
||||
SDL_SetError("Invalid file seek");
|
||||
return -1;
|
||||
}
|
||||
|
||||
static int read(SDL_RWops *context, void *ptr, int size, int maxnum)
|
||||
{
|
||||
RWSubFile *self = static_cast<RWSubFile*>(context->hidden.unknown.data1);
|
||||
return fread(ptr, size, maxnum, self->fp);
|
||||
}
|
||||
|
||||
static int write(SDL_RWops *context, const void *ptr, int size, int num)
|
||||
{
|
||||
RWSubFile *self = static_cast<RWSubFile*>(context->hidden.unknown.data1);
|
||||
return fwrite(ptr, size, num, self->fp);
|
||||
}
|
||||
|
||||
static int close(SDL_RWops *context)
|
||||
{
|
||||
RWSubFile *self = static_cast<RWSubFile*>(context->hidden.unknown.data1);
|
||||
if(context->type != 0xdeadbeef)
|
||||
{
|
||||
SDL_SetError("Wrong kind of RWops for RWSubfile::close");
|
||||
return -1;
|
||||
}
|
||||
|
||||
delete self;
|
||||
SDL_FreeRW(context);
|
||||
|
||||
return 0;
|
||||
}
|
||||
};
|
||||
|
||||
|
||||
static ALenum checkALError(const char *fn, unsigned int ln)
|
||||
{
|
||||
ALenum err = alGetError();
|
||||
if(err != AL_NO_ERROR)
|
||||
{
|
||||
if(strchr(fn, '/'))
|
||||
fn = strrchr(fn, '/')+1;
|
||||
else if(strchr(fn, '\\'))
|
||||
fn = strrchr(fn, '\\')+1;
|
||||
Printf(">>>>>>>>>>>> Received AL error %s (%#x), %s:%u\n", alGetString(err), err, fn, ln);
|
||||
}
|
||||
return err;
|
||||
}
|
||||
#define getALError() checkALError(__FILE__, __LINE__)
|
||||
|
||||
|
||||
bool OpenALSoundStream::SetupSource()
|
||||
{
|
||||
if(Renderer->FreeSfx.size() == 0)
|
||||
{
|
||||
FSoundChan *lowest = Renderer->FindLowestChannel();
|
||||
if(lowest) Renderer->StopChannel(lowest);
|
||||
|
||||
if(Renderer->FreeSfx.size() == 0)
|
||||
return false;
|
||||
}
|
||||
Source = Renderer->FreeSfx.back();
|
||||
Renderer->FreeSfx.pop_back();
|
||||
|
||||
alSource3f(Source, AL_DIRECTION, 0.f, 0.f, 0.f);
|
||||
alSource3f(Source, AL_VELOCITY, 0.f, 0.f, 0.f);
|
||||
alSource3f(Source, AL_POSITION, 0.f, 0.f, 0.f);
|
||||
alSourcef(Source, AL_MAX_GAIN, Renderer->MusicVolume);
|
||||
alSourcef(Source, AL_GAIN, Volume*Renderer->MusicVolume);
|
||||
alSourcef(Source, AL_PITCH, 1.f);
|
||||
alSourcef(Source, AL_ROLLOFF_FACTOR, 0.f);
|
||||
alSourcef(Source, AL_SEC_OFFSET, 0.f);
|
||||
alSourcei(Source, AL_SOURCE_RELATIVE, AL_TRUE);
|
||||
alSourcei(Source, AL_LOOPING, AL_FALSE);
|
||||
if(Renderer->EnvSlot)
|
||||
{
|
||||
alSourcef(Source, AL_ROOM_ROLLOFF_FACTOR, 0.f);
|
||||
alSourcef(Source, AL_AIR_ABSORPTION_FACTOR, 0.f);
|
||||
alSourcei(Source, AL_DIRECT_FILTER, AL_FILTER_NULL);
|
||||
alSource3i(Source, AL_AUXILIARY_SEND_FILTER, 0, 0, AL_FILTER_NULL);
|
||||
}
|
||||
|
||||
alGenBuffers(Buffers.size(), &Buffers[0]);
|
||||
return (getALError() == AL_NO_ERROR);
|
||||
}
|
||||
|
||||
OpenALSoundStream::OpenALSoundStream(OpenALSoundRenderer *renderer)
|
||||
: Renderer(renderer), Sample(NULL), Source(0), Playing(false),
|
||||
Looping(false), Buffers(4), SampleRate(0), Format(0),
|
||||
NeedSwab(false), Volume(1.f)
|
||||
{
|
||||
for(size_t i = 0;i < Buffers.size();i++)
|
||||
Buffers[i] = 0;
|
||||
Renderer->Streams.push_back(this);
|
||||
}
|
||||
|
||||
OpenALSoundStream::~OpenALSoundStream()
|
||||
{
|
||||
Playing = false;
|
||||
|
||||
Sound_FreeSample(Sample);
|
||||
Sample = NULL;
|
||||
|
||||
if(Source)
|
||||
{
|
||||
alSourceRewind(Source);
|
||||
alSourcei(Source, AL_BUFFER, 0);
|
||||
|
||||
Renderer->FreeSfx.push_back(Source);
|
||||
Source = 0;
|
||||
}
|
||||
|
||||
if(Buffers.size() > 0)
|
||||
{
|
||||
alDeleteBuffers(Buffers.size(), &Buffers[0]);
|
||||
Buffers.clear();
|
||||
}
|
||||
getALError();
|
||||
|
||||
Renderer->Streams.erase(std::find(Renderer->Streams.begin(),
|
||||
Renderer->Streams.end(), this));
|
||||
Renderer = NULL;
|
||||
}
|
||||
|
||||
bool OpenALSoundStream::Play(bool looping, float vol)
|
||||
{
|
||||
if(Playing)
|
||||
return true;
|
||||
|
||||
alSourceRewind(Source);
|
||||
alSourcei(Source, AL_BUFFER, 0);
|
||||
|
||||
Looping = looping;
|
||||
SetVolume(vol);
|
||||
|
||||
for(size_t i = 0;i < Buffers.size();i++)
|
||||
{
|
||||
size_t count = Sound_Decode(Sample);
|
||||
if(count == 0 && Looping)
|
||||
{
|
||||
Sound_Seek(Sample, 0);
|
||||
count = Sound_Decode(Sample);
|
||||
}
|
||||
alBufferData(Buffers[i], Format, GetData(count), count, SampleRate);
|
||||
}
|
||||
Playing = (getALError() == AL_NO_ERROR);
|
||||
if(Playing)
|
||||
{
|
||||
alSourceQueueBuffers(Source, Buffers.size(), &Buffers[0]);
|
||||
alSourcePlay(Source);
|
||||
Playing = (getALError() == AL_NO_ERROR);
|
||||
}
|
||||
return Playing;
|
||||
}
|
||||
|
||||
void OpenALSoundStream::Stop()
|
||||
{
|
||||
alSourceRewind(Source);
|
||||
alSourcei(Source, AL_BUFFER, 0);
|
||||
}
|
||||
|
||||
bool OpenALSoundStream::SetPaused(bool paused)
|
||||
{
|
||||
if(paused) alSourcePause(Source);
|
||||
else alSourcePlay(Source);
|
||||
return (getALError() == AL_NO_ERROR);
|
||||
}
|
||||
|
||||
void OpenALSoundStream::SetVolume(float vol)
|
||||
{
|
||||
if(vol >= 0.f) Volume = vol;
|
||||
alSourcef(Source, AL_GAIN, Volume*Renderer->MusicVolume);
|
||||
}
|
||||
|
||||
unsigned int OpenALSoundStream::GetPosition()
|
||||
{
|
||||
return 0;
|
||||
}
|
||||
|
||||
bool OpenALSoundStream::SetPosition(unsigned int val)
|
||||
{
|
||||
return false;
|
||||
}
|
||||
|
||||
bool OpenALSoundStream::IsEnded()
|
||||
{
|
||||
if(!Playing)
|
||||
return true;
|
||||
|
||||
ALint processed, state, queued;
|
||||
alGetSourcei(Source, AL_SOURCE_STATE, &state);
|
||||
alGetSourcei(Source, AL_BUFFERS_QUEUED, &queued);
|
||||
alGetSourcei(Source, AL_BUFFERS_PROCESSED, &processed);
|
||||
while(processed-- > 0)
|
||||
{
|
||||
ALuint buf = 0;
|
||||
alSourceUnqueueBuffers(Source, 1, &buf);
|
||||
queued--;
|
||||
|
||||
size_t count = Sound_Decode(Sample);
|
||||
if(count == 0 && Looping)
|
||||
{
|
||||
Sound_Seek(Sample, 0);
|
||||
count = Sound_Decode(Sample);
|
||||
}
|
||||
if(count > 0)
|
||||
{
|
||||
alBufferData(buf, Format, GetData(count), count, SampleRate);
|
||||
alSourceQueueBuffers(Source, 1, &buf);
|
||||
queued++;
|
||||
}
|
||||
}
|
||||
Playing = (getALError() == AL_NO_ERROR && queued > 0);
|
||||
if(Playing && state != AL_PLAYING && state != AL_PAUSED)
|
||||
{
|
||||
alSourcePlay(Source);
|
||||
Playing = (getALError() == AL_NO_ERROR);
|
||||
}
|
||||
|
||||
return !Playing;
|
||||
}
|
||||
|
||||
void *OpenALSoundStream::GetData(size_t bytes)
|
||||
{
|
||||
void *data = Sample->buffer;
|
||||
if(NeedSwab)
|
||||
{
|
||||
short *samples = reinterpret_cast<short*>(data);
|
||||
size_t count = bytes >> 1;
|
||||
for(size_t i = 0;i < count;i++)
|
||||
{
|
||||
short smp = *samples;
|
||||
*(samples++) = ((smp>>8)&0x00FF) | ((smp<<8)*0xFF00);
|
||||
}
|
||||
}
|
||||
return data;
|
||||
}
|
||||
|
||||
bool OpenALSoundStream::InitSample()
|
||||
{
|
||||
UInt32 smpsize = 0;
|
||||
SampleRate = Sample->actual.rate;
|
||||
|
||||
Format = AL_NONE;
|
||||
if(Sample->actual.format == AUDIO_U8)
|
||||
{
|
||||
if(Sample->actual.channels == 1)
|
||||
Format = AL_FORMAT_MONO8;
|
||||
else if(Sample->actual.channels == 2)
|
||||
Format = AL_FORMAT_STEREO8;
|
||||
smpsize = 1 * Sample->actual.channels;
|
||||
}
|
||||
else if(Sample->actual.format == AUDIO_S16LSB || Sample->actual.format == AUDIO_S16MSB)
|
||||
{
|
||||
NeedSwab = (Sample->actual.format != AUDIO_S16SYS);
|
||||
if(Sample->actual.channels == 1)
|
||||
Format = AL_FORMAT_MONO16;
|
||||
else if(Sample->actual.channels == 2)
|
||||
Format = AL_FORMAT_STEREO16;
|
||||
smpsize = 1 * Sample->actual.channels;
|
||||
}
|
||||
|
||||
if(Format == AL_NONE)
|
||||
{
|
||||
Printf("Unsupported sound format (0x%04x, %d channels)\n", Sample->actual.format, Sample->actual.channels);
|
||||
return false;
|
||||
}
|
||||
|
||||
Uint32 bufsize = (UInt32)(BufferTime*SampleRate) * smpsize;
|
||||
if(Sound_SetBufferSize(Sample, bufsize) == 0)
|
||||
{
|
||||
Printf("Failed to set buffer size to %u bytes: %s\n", bufsize, Sound_GetError());
|
||||
return false;
|
||||
}
|
||||
|
||||
return true;
|
||||
}
|
||||
|
||||
bool OpenALSoundStream::Init(const char *filename, int offset, int length)
|
||||
{
|
||||
if(!SetupSource())
|
||||
return false;
|
||||
|
||||
if(offset == 0)
|
||||
Sample = Sound_NewSampleFromFile(filename, NULL, 0);
|
||||
else
|
||||
{
|
||||
FILE *fp = fopen(filename, "rb");
|
||||
if(!fp)
|
||||
{
|
||||
Printf("Failed to open %s\n", filename);
|
||||
return false;
|
||||
}
|
||||
|
||||
const char *ext = strrchr(filename, '.');
|
||||
if(ext) ext++;
|
||||
|
||||
SDL_RWops *ops = SDL_AllocRW();
|
||||
ops->seek = RWSubFile::seek;
|
||||
ops->read = RWSubFile::read;
|
||||
ops->write = RWSubFile::write;
|
||||
ops->close = RWSubFile::close;
|
||||
ops->type = 0xdeadbeef;
|
||||
ops->hidden.unknown.data1 = new RWSubFile(fp, offset, length);
|
||||
|
||||
Sample = Sound_NewSample(ops, ext, NULL, 0);
|
||||
}
|
||||
if(!Sample)
|
||||
{
|
||||
Printf("Could not open audio in %s (%s)\n", filename, Sound_GetError());
|
||||
return false;
|
||||
}
|
||||
|
||||
return InitSample();
|
||||
}
|
||||
|
||||
bool OpenALSoundStream::Init(const BYTE *data, unsigned int datalen)
|
||||
{
|
||||
if(!SetupSource())
|
||||
return false;
|
||||
|
||||
Sample = Sound_NewSample(SDL_RWFromConstMem(data, datalen), NULL, NULL, 0);
|
||||
if(!Sample)
|
||||
{
|
||||
Printf("Could not read audio: %s\n", Sound_GetError());
|
||||
return false;
|
||||
}
|
||||
|
||||
return InitSample();
|
||||
}
|
||||
|
||||
|
||||
Decoder::Decoder(const void* data, unsigned int datalen)
|
||||
: Sample(NULL), NeedSwab(false)
|
||||
{
|
||||
Sample = Sound_NewSample(SDL_RWFromConstMem(data, datalen), NULL, NULL, 65536);
|
||||
}
|
||||
|
||||
|
||||
Decoder::~Decoder()
|
||||
{
|
||||
Sound_FreeSample(Sample);
|
||||
Sample = NULL;
|
||||
}
|
||||
|
||||
bool Decoder::GetFormat(ALenum *format, ALuint *rate)
|
||||
{
|
||||
ALenum fmt = AL_NONE;
|
||||
if(Sample->actual.format == AUDIO_U8)
|
||||
{
|
||||
if(Sample->actual.channels == 1)
|
||||
fmt = AL_FORMAT_MONO8;
|
||||
else if(Sample->actual.channels == 2)
|
||||
fmt = AL_FORMAT_STEREO8;
|
||||
}
|
||||
else if(Sample->actual.format == AUDIO_S16LSB || Sample->actual.format == AUDIO_S16MSB)
|
||||
{
|
||||
NeedSwab = (Sample->actual.format != AUDIO_S16SYS);
|
||||
if(Sample->actual.channels == 1)
|
||||
fmt = AL_FORMAT_MONO16;
|
||||
else if(Sample->actual.channels == 2)
|
||||
fmt = AL_FORMAT_STEREO16;
|
||||
}
|
||||
|
||||
if(fmt == AL_NONE)
|
||||
{
|
||||
Printf("Unsupported sound format (0x%04x, %d channels)\n", Sample->actual.format, Sample->actual.channels);
|
||||
return false;
|
||||
}
|
||||
|
||||
*format = fmt;
|
||||
*rate = Sample->actual.rate;
|
||||
return true;
|
||||
}
|
||||
|
||||
void* Decoder::GetData(ALsizei *size)
|
||||
{
|
||||
UInt32 got = Sound_DecodeAll(Sample);
|
||||
if(got == 0)
|
||||
{
|
||||
*size = 0;
|
||||
return NULL;
|
||||
}
|
||||
|
||||
void *data = Sample->buffer;
|
||||
if(NeedSwab)
|
||||
{
|
||||
short *samples = reinterpret_cast<short*>(data);
|
||||
size_t count = got >> 1;
|
||||
for(size_t i = 0;i < count;i++)
|
||||
{
|
||||
short smp = *samples;
|
||||
*(samples++) = ((smp>>8)&0x00FF) | ((smp<<8)*0xFF00);
|
||||
}
|
||||
}
|
||||
|
||||
*size = got;
|
||||
return data;
|
||||
}
|
66
src/sound/oalsdlsound.h
Normal file
66
src/sound/oalsdlsound.h
Normal file
|
@ -0,0 +1,66 @@
|
|||
#ifndef OALSDLSOUND_H
|
||||
#define OALSDLSOUND_H
|
||||
|
||||
#include "oalsound.h"
|
||||
#include "tempfiles.h"
|
||||
|
||||
#include "SDL_sound.h"
|
||||
|
||||
class OpenALSoundStream : public SoundStream
|
||||
{
|
||||
OpenALSoundRenderer *Renderer;
|
||||
Sound_Sample *Sample;
|
||||
ALuint Source;
|
||||
|
||||
bool Playing;
|
||||
bool Looping;
|
||||
|
||||
static const ALfloat BufferTime = 0.2f;
|
||||
std::vector<ALuint> Buffers;
|
||||
|
||||
ALuint SampleRate;
|
||||
ALenum Format;
|
||||
|
||||
bool NeedSwab;
|
||||
void *GetData(size_t bytes);
|
||||
|
||||
// General methods
|
||||
bool SetupSource();
|
||||
bool InitSample();
|
||||
|
||||
ALfloat Volume;
|
||||
|
||||
public:
|
||||
OpenALSoundStream(OpenALSoundRenderer *renderer);
|
||||
virtual ~OpenALSoundStream();
|
||||
|
||||
virtual bool Play(bool looping, float vol);
|
||||
virtual void Stop();
|
||||
virtual bool SetPaused(bool paused);
|
||||
|
||||
virtual void SetVolume(float vol);
|
||||
|
||||
virtual unsigned int GetPosition();
|
||||
virtual bool SetPosition(unsigned int val);
|
||||
|
||||
virtual bool IsEnded();
|
||||
|
||||
bool Init(const char *filename, int offset, int length);
|
||||
bool Init(const BYTE *data, unsigned int datalen);
|
||||
};
|
||||
|
||||
|
||||
class Decoder
|
||||
{
|
||||
Sound_Sample *Sample;
|
||||
bool NeedSwab;
|
||||
|
||||
public:
|
||||
Decoder(const void *data, unsigned int datalen);
|
||||
virtual ~Decoder();
|
||||
|
||||
bool GetFormat(ALenum *format, ALuint *rate);
|
||||
void *GetData(ALsizei *size);
|
||||
};
|
||||
|
||||
#endif /* OALSDLSOUND_H */
|
|
@ -94,6 +94,9 @@ void I_BuildALDeviceList(FOptionValues *opt)
|
|||
#include <string>
|
||||
#include <vector>
|
||||
|
||||
#ifdef WITH_SDL_SOUND
|
||||
#include "SDL_sound.h"
|
||||
#endif
|
||||
|
||||
EXTERN_CVAR (Int, snd_channels)
|
||||
EXTERN_CVAR (Int, snd_samplerate)
|
||||
|
@ -137,6 +140,315 @@ static ALCenum checkALCError(ALCdevice *device, const char *fn, unsigned int ln)
|
|||
}
|
||||
#define getALCError(d) checkALCError((d), __FILE__, __LINE__)
|
||||
|
||||
#ifdef WITH_SDL_SOUND
|
||||
#include "oalsdlsound.h"
|
||||
#else
|
||||
|
||||
class OpenALSoundStream : public SoundStream
|
||||
{
|
||||
OpenALSoundRenderer *Renderer;
|
||||
|
||||
public:
|
||||
ALfloat Volume;
|
||||
|
||||
OpenALSoundStream(OpenALSoundRenderer *renderer)
|
||||
: Renderer(renderer), Volume(1.0f)
|
||||
{ Renderer->Streams.push_back(this); }
|
||||
|
||||
virtual ~OpenALSoundStream()
|
||||
{
|
||||
Renderer->Streams.erase(std::find(Renderer->Streams.begin(),
|
||||
Renderer->Streams.end(), this));
|
||||
Renderer = NULL;
|
||||
}
|
||||
|
||||
|
||||
virtual bool Play(bool, float)
|
||||
{ return false; }
|
||||
|
||||
virtual void Stop()
|
||||
{ }
|
||||
|
||||
virtual void SetVolume(float vol)
|
||||
{ Volume = vol; }
|
||||
|
||||
virtual bool SetPaused(bool)
|
||||
{ return false; }
|
||||
|
||||
virtual unsigned int GetPosition()
|
||||
{ return 0; }
|
||||
|
||||
virtual bool IsEnded()
|
||||
{ return true; }
|
||||
|
||||
bool Init(const char*)
|
||||
{ return false; }
|
||||
|
||||
bool Init(const BYTE*, unsigned int)
|
||||
{ return false; }
|
||||
};
|
||||
|
||||
class Decoder
|
||||
{
|
||||
public:
|
||||
Decoder(const void*, unsigned int) { }
|
||||
virtual ~Decoder() { }
|
||||
|
||||
bool GetFormat(ALenum*, ALuint*)
|
||||
{ return false; }
|
||||
void *GetData(ALsizei *size)
|
||||
{ *size = 0; return NULL; }
|
||||
};
|
||||
#endif
|
||||
|
||||
class OpenALCallbackStream : public SoundStream
|
||||
{
|
||||
OpenALSoundRenderer *Renderer;
|
||||
|
||||
SoundStreamCallback Callback;
|
||||
void *UserData;
|
||||
|
||||
std::vector<ALubyte> Data;
|
||||
|
||||
ALsizei SampleRate;
|
||||
ALenum Format;
|
||||
|
||||
static const int BufferCount = 4;
|
||||
ALuint Buffers[BufferCount];
|
||||
ALuint Source;
|
||||
|
||||
bool Playing;
|
||||
|
||||
bool SetupSource()
|
||||
{
|
||||
/* Get a source, killing the farthest, lowest-priority sound if needed */
|
||||
if(Renderer->FreeSfx.size() == 0)
|
||||
{
|
||||
FSoundChan *lowest = Renderer->FindLowestChannel();
|
||||
if(lowest) Renderer->StopChannel(lowest);
|
||||
|
||||
if(Renderer->FreeSfx.size() == 0)
|
||||
return false;
|
||||
}
|
||||
Source = Renderer->FreeSfx.back();
|
||||
Renderer->FreeSfx.pop_back();
|
||||
|
||||
/* Set the default properties for localized playback */
|
||||
alSource3f(Source, AL_DIRECTION, 0.f, 0.f, 0.f);
|
||||
alSource3f(Source, AL_VELOCITY, 0.f, 0.f, 0.f);
|
||||
alSource3f(Source, AL_POSITION, 0.f, 0.f, 0.f);
|
||||
alSourcef(Source, AL_MAX_GAIN, 1.f);
|
||||
alSourcef(Source, AL_GAIN, 1.f);
|
||||
alSourcef(Source, AL_PITCH, 1.f);
|
||||
alSourcef(Source, AL_ROLLOFF_FACTOR, 0.f);
|
||||
alSourcef(Source, AL_SEC_OFFSET, 0.f);
|
||||
alSourcei(Source, AL_SOURCE_RELATIVE, AL_TRUE);
|
||||
alSourcei(Source, AL_LOOPING, AL_FALSE);
|
||||
if(Renderer->EnvSlot)
|
||||
{
|
||||
alSourcef(Source, AL_ROOM_ROLLOFF_FACTOR, 0.f);
|
||||
alSourcef(Source, AL_AIR_ABSORPTION_FACTOR, 0.f);
|
||||
alSourcei(Source, AL_DIRECT_FILTER, AL_FILTER_NULL);
|
||||
alSource3i(Source, AL_AUXILIARY_SEND_FILTER, 0, 0, AL_FILTER_NULL);
|
||||
}
|
||||
|
||||
alGenBuffers(BufferCount, Buffers);
|
||||
return (getALError() == AL_NO_ERROR);
|
||||
}
|
||||
|
||||
public:
|
||||
ALfloat Volume;
|
||||
|
||||
OpenALCallbackStream(OpenALSoundRenderer *renderer)
|
||||
: Renderer(renderer), Source(0), Playing(false), Volume(1.0f)
|
||||
{
|
||||
Renderer->Streams.push_back(this);
|
||||
memset(Buffers, 0, sizeof(Buffers));
|
||||
}
|
||||
|
||||
virtual ~OpenALCallbackStream()
|
||||
{
|
||||
if(Source)
|
||||
{
|
||||
alSourceRewind(Source);
|
||||
alSourcei(Source, AL_BUFFER, 0);
|
||||
|
||||
Renderer->FreeSfx.push_back(Source);
|
||||
Source = 0;
|
||||
}
|
||||
|
||||
if(Buffers[0])
|
||||
{
|
||||
alDeleteBuffers(BufferCount, &Buffers[0]);
|
||||
memset(Buffers, 0, sizeof(Buffers));
|
||||
}
|
||||
getALError();
|
||||
|
||||
Renderer->Streams.erase(std::find(Renderer->Streams.begin(),
|
||||
Renderer->Streams.end(), this));
|
||||
Renderer = NULL;
|
||||
}
|
||||
|
||||
|
||||
virtual bool Play(bool loop, float vol)
|
||||
{
|
||||
SetVolume(vol);
|
||||
|
||||
if(Playing)
|
||||
return true;
|
||||
|
||||
/* Clear the buffer queue, then fill and queue each buffer */
|
||||
alSourcei(Source, AL_BUFFER, 0);
|
||||
for(int i = 0;i < BufferCount;i++)
|
||||
{
|
||||
if(!Callback(this, &Data[0], Data.size(), UserData))
|
||||
{
|
||||
if(i == 0)
|
||||
return false;
|
||||
break;
|
||||
}
|
||||
|
||||
alBufferData(Buffers[i], Format, &Data[0], Data.size(), SampleRate);
|
||||
alSourceQueueBuffers(Source, 1, &Buffers[i]);
|
||||
}
|
||||
if(getALError() != AL_NO_ERROR)
|
||||
return false;
|
||||
|
||||
alSourcePlay(Source);
|
||||
Playing = (getALError()==AL_NO_ERROR);
|
||||
|
||||
return Playing;
|
||||
}
|
||||
|
||||
virtual void Stop()
|
||||
{
|
||||
if(!Playing)
|
||||
return;
|
||||
|
||||
alSourceStop(Source);
|
||||
alSourcei(Source, AL_BUFFER, 0);
|
||||
getALError();
|
||||
|
||||
Playing = false;
|
||||
}
|
||||
|
||||
virtual void SetVolume(float vol)
|
||||
{
|
||||
if(vol >= 0.0f) Volume = vol;
|
||||
alSourcef(Source, AL_GAIN, Renderer->MusicVolume*Volume);
|
||||
getALError();
|
||||
}
|
||||
|
||||
virtual bool SetPaused(bool pause)
|
||||
{
|
||||
if(pause)
|
||||
alSourcePause(Source);
|
||||
else
|
||||
alSourcePlay(Source);
|
||||
return (getALError()==AL_NO_ERROR);
|
||||
}
|
||||
|
||||
virtual unsigned int GetPosition()
|
||||
{ return 0; }
|
||||
|
||||
virtual bool IsEnded()
|
||||
{
|
||||
if(!Playing)
|
||||
return true;
|
||||
|
||||
ALint state, processed;
|
||||
alGetSourcei(Source, AL_SOURCE_STATE, &state);
|
||||
alGetSourcei(Source, AL_BUFFERS_PROCESSED, &processed);
|
||||
|
||||
Playing = (getALError()==AL_NO_ERROR);
|
||||
if(!Playing)
|
||||
return true;
|
||||
|
||||
// For each processed buffer in the queue...
|
||||
while(processed > 0)
|
||||
{
|
||||
ALuint bufid;
|
||||
|
||||
// Unqueue the oldest buffer, fill it with more data, and queue it
|
||||
// on the end
|
||||
alSourceUnqueueBuffers(Source, 1, &bufid);
|
||||
processed--;
|
||||
|
||||
if(Callback(this, &Data[0], Data.size(), UserData))
|
||||
{
|
||||
alBufferData(bufid, Format, &Data[0], Data.size(), SampleRate);
|
||||
alSourceQueueBuffers(Source, 1, &bufid);
|
||||
}
|
||||
}
|
||||
|
||||
// If the source is not playing or paused, and there are buffers queued,
|
||||
// then there was an underrun. Restart the source.
|
||||
Playing = (getALError()==AL_NO_ERROR);
|
||||
if(Playing && state != AL_PLAYING && state != AL_PAUSED)
|
||||
{
|
||||
ALint queued = 0;
|
||||
alGetSourcei(Source, AL_BUFFERS_QUEUED, &queued);
|
||||
|
||||
Playing = (getALError() == AL_NO_ERROR) && (queued > 0);
|
||||
if(Playing)
|
||||
{
|
||||
alSourcePlay(Source);
|
||||
Playing = (getALError()==AL_NO_ERROR);
|
||||
}
|
||||
}
|
||||
|
||||
return !Playing;
|
||||
}
|
||||
|
||||
bool Init(SoundStreamCallback callback, int buffbytes, int flags, int samplerate, void *userdata)
|
||||
{
|
||||
if(!SetupSource())
|
||||
return false;
|
||||
|
||||
Callback = callback;
|
||||
UserData = userdata;
|
||||
SampleRate = samplerate;
|
||||
|
||||
Format = AL_NONE;
|
||||
if((flags&Bits8)) /* Signed or unsigned? We assume unsigned 8-bit... */
|
||||
{
|
||||
if((flags&Mono)) Format = AL_FORMAT_MONO8;
|
||||
else Format = AL_FORMAT_STEREO8;
|
||||
}
|
||||
else if(!(flags&(Bits32|Float)))
|
||||
{
|
||||
if((flags&Mono)) Format = AL_FORMAT_MONO16;
|
||||
else Format = AL_FORMAT_STEREO16;
|
||||
}
|
||||
|
||||
if(Format == AL_NONE)
|
||||
{
|
||||
Printf("Unsupported format: 0x%x\n", flags);
|
||||
return false;
|
||||
}
|
||||
|
||||
int smpsize = 1;
|
||||
if((flags&Bits8))
|
||||
smpsize *= 1;
|
||||
else if((flags&(Bits32|Float)))
|
||||
smpsize *= 4;
|
||||
else
|
||||
smpsize *= 2;
|
||||
|
||||
if((flags&Mono))
|
||||
smpsize *= 1;
|
||||
else
|
||||
smpsize *= 2;
|
||||
|
||||
buffbytes += smpsize-1;
|
||||
buffbytes -= buffbytes%smpsize;
|
||||
Data.resize(buffbytes);
|
||||
|
||||
return true;
|
||||
}
|
||||
};
|
||||
|
||||
|
||||
extern ReverbContainer *ForcedEnvironment;
|
||||
|
||||
#define PITCH_MULT (0.7937005f) /* Approx. 4 semitones lower; what Nash suggested */
|
||||
|
@ -198,72 +510,6 @@ static ALenum FormatFromDesc(int bits, int channels)
|
|||
}
|
||||
|
||||
|
||||
class OpenALSoundStream : public SoundStream
|
||||
{
|
||||
OpenALSoundRenderer *Renderer;
|
||||
|
||||
public:
|
||||
FTempFileName tmpfile;
|
||||
ALfloat Volume;
|
||||
|
||||
OpenALSoundStream(OpenALSoundRenderer *renderer)
|
||||
: Renderer(renderer), Volume(1.0f)
|
||||
{ Renderer->Streams.push_back(this); }
|
||||
|
||||
virtual ~OpenALSoundStream()
|
||||
{
|
||||
Renderer->Streams.erase(std::find(Renderer->Streams.begin(),
|
||||
Renderer->Streams.end(), this));
|
||||
Renderer = NULL;
|
||||
}
|
||||
|
||||
|
||||
virtual bool Play(bool, float)
|
||||
{ return false; }
|
||||
|
||||
virtual void Stop()
|
||||
{ }
|
||||
|
||||
virtual void SetVolume(float vol)
|
||||
{ Volume = vol; }
|
||||
|
||||
virtual bool SetPaused(bool)
|
||||
{ return false; }
|
||||
|
||||
virtual unsigned int GetPosition()
|
||||
{ return 0; }
|
||||
|
||||
virtual bool IsEnded()
|
||||
{ return true; }
|
||||
|
||||
bool Init(const char*)
|
||||
{ return false; }
|
||||
|
||||
bool Init(const BYTE*, unsigned int)
|
||||
{ return false; }
|
||||
|
||||
bool Init(SoundStreamCallback, int, int, int, void*)
|
||||
{ return false; }
|
||||
};
|
||||
|
||||
class Decoder
|
||||
{
|
||||
public:
|
||||
std::vector<BYTE> OutData;
|
||||
ALint LoopPts[2];
|
||||
ALsizei OutRate;
|
||||
ALuint OutChannels;
|
||||
ALuint OutBits;
|
||||
|
||||
Decoder()
|
||||
: OutRate(0), OutChannels(0), OutBits(0)
|
||||
{ LoopPts[0] = LoopPts[1] = 0; }
|
||||
|
||||
bool Decode(const void*, unsigned int, int=0)
|
||||
{ return false; }
|
||||
};
|
||||
|
||||
|
||||
template<typename T>
|
||||
static void LoadALFunc(const char *name, T *x)
|
||||
{ *x = reinterpret_cast<T>(alGetProcAddress(name)); }
|
||||
|
@ -276,6 +522,19 @@ OpenALSoundRenderer::OpenALSoundRenderer()
|
|||
|
||||
Printf("I_InitSound: Initializing OpenAL\n");
|
||||
|
||||
#ifdef WITH_SDL_SOUND
|
||||
static bool sdl_sound_inited = false;
|
||||
if(!sdl_sound_inited)
|
||||
{
|
||||
if(Sound_Init() == 0)
|
||||
{
|
||||
Printf(TEXTCOLOR_RED" Failed to init SDL_sound: %s\n", Sound_GetError());
|
||||
return;
|
||||
}
|
||||
sdl_sound_inited = true;
|
||||
}
|
||||
#endif
|
||||
|
||||
if(strcmp(snd_aldevice, "Default") != 0)
|
||||
{
|
||||
Device = alcOpenDevice(*snd_aldevice);
|
||||
|
@ -538,8 +797,8 @@ void OpenALSoundRenderer::SetSfxVolume(float volume)
|
|||
void OpenALSoundRenderer::SetMusicVolume(float volume)
|
||||
{
|
||||
MusicVolume = volume;
|
||||
foreach(OpenALSoundStream*, i, Streams)
|
||||
(*i)->SetVolume((*i)->Volume);
|
||||
foreach(SoundStream*, i, Streams)
|
||||
(*i)->SetVolume(-1.f);
|
||||
}
|
||||
|
||||
unsigned int OpenALSoundRenderer::GetMSLength(SoundHandle sfx)
|
||||
|
@ -656,26 +915,23 @@ SoundHandle OpenALSoundRenderer::LoadSoundRaw(BYTE *sfxdata, int length, int fre
|
|||
SoundHandle OpenALSoundRenderer::LoadSound(BYTE *sfxdata, int length)
|
||||
{
|
||||
SoundHandle retval = { NULL };
|
||||
ALenum format;
|
||||
ALuint srate;
|
||||
|
||||
Decoder decoder;
|
||||
if(!decoder.Decode(sfxdata, length))
|
||||
{
|
||||
DPrintf("Failed to decode sound\n");
|
||||
Decoder decoder(sfxdata, length);
|
||||
if(!decoder.GetFormat(&format, &srate))
|
||||
return retval;
|
||||
}
|
||||
|
||||
ALenum format = FormatFromDesc(decoder.OutBits, decoder.OutChannels);
|
||||
if(format == AL_NONE)
|
||||
{
|
||||
Printf("Unhandled format: %d bit, %d channel\n", decoder.OutBits, decoder.OutChannels);
|
||||
ALsizei size;
|
||||
void *data = decoder.GetData(&size);
|
||||
if(data == NULL)
|
||||
return retval;
|
||||
}
|
||||
|
||||
ALenum err;
|
||||
ALuint buffer = 0;
|
||||
alGenBuffers(1, &buffer);
|
||||
alBufferData(buffer, format, &decoder.OutData[0],
|
||||
decoder.OutData.size(), decoder.OutRate);
|
||||
alBufferData(buffer, format, data, size, srate);
|
||||
|
||||
ALenum err;
|
||||
if((err=getALError()) != AL_NO_ERROR)
|
||||
{
|
||||
Printf("Failed to buffer data: %s\n", alGetString(err));
|
||||
|
@ -684,19 +940,6 @@ SoundHandle OpenALSoundRenderer::LoadSound(BYTE *sfxdata, int length)
|
|||
return retval;
|
||||
}
|
||||
|
||||
if(LoopPoints && decoder.LoopPts[1] > decoder.LoopPts[0])
|
||||
{
|
||||
alBufferiv(buffer, AL_LOOP_POINTS_SOFT, decoder.LoopPts);
|
||||
getALError();
|
||||
}
|
||||
else if(decoder.LoopPts[1] > decoder.LoopPts[0])
|
||||
{
|
||||
static bool warned = false;
|
||||
if(!warned)
|
||||
Printf("Loop points not supported!\n");
|
||||
warned = true;
|
||||
}
|
||||
|
||||
retval.data = new ALuint(buffer);
|
||||
return retval;
|
||||
}
|
||||
|
@ -731,74 +974,43 @@ void OpenALSoundRenderer::UnloadSound(SoundHandle sfx)
|
|||
|
||||
short *OpenALSoundRenderer::DecodeSample(int outlen, const void *coded, int sizebytes, ECodecType type)
|
||||
{
|
||||
Decoder decoder;
|
||||
// Force 16-bit
|
||||
if(!decoder.Decode(coded, sizebytes, 16))
|
||||
short *samples = (short*)malloc(outlen);
|
||||
memset(samples, 0, outlen);
|
||||
|
||||
Decoder decoder(coded, sizebytes);
|
||||
ALenum format;
|
||||
ALuint srate;
|
||||
|
||||
if(!decoder.GetFormat(&format, &srate))
|
||||
return samples;
|
||||
if(format != AL_FORMAT_MONO16)
|
||||
{
|
||||
DPrintf("Failed to decode sample\n");
|
||||
return NULL;
|
||||
}
|
||||
if(decoder.OutChannels != 1)
|
||||
{
|
||||
DPrintf("Sample is not mono\n");
|
||||
return NULL;
|
||||
DPrintf("Sample is not 16-bit mono\n");
|
||||
return samples;
|
||||
}
|
||||
|
||||
short *samples = (short*)malloc(outlen);
|
||||
if((size_t)outlen > decoder.OutData.size())
|
||||
{
|
||||
memcpy(samples, &decoder.OutData[0], decoder.OutData.size());
|
||||
memset(&samples[decoder.OutData.size()/sizeof(short)], 0, outlen-decoder.OutData.size());
|
||||
}
|
||||
else
|
||||
memcpy(samples, &decoder.OutData[0], outlen);
|
||||
ALsizei size;
|
||||
void *data = decoder.GetData(&size);
|
||||
if(data != NULL)
|
||||
memcpy(samples, data, std::min<size_t>(size, outlen));
|
||||
|
||||
return samples;
|
||||
}
|
||||
|
||||
SoundStream *OpenALSoundRenderer::CreateStream(SoundStreamCallback callback, int buffbytes, int flags, int samplerate, void *userdata)
|
||||
{
|
||||
OpenALSoundStream *stream = new OpenALSoundStream(this);
|
||||
std::auto_ptr<OpenALCallbackStream> stream(new OpenALCallbackStream(this));
|
||||
if(!stream->Init(callback, buffbytes, flags, samplerate, userdata))
|
||||
{
|
||||
delete stream;
|
||||
stream = NULL;
|
||||
}
|
||||
return stream;
|
||||
return NULL;
|
||||
return stream.release();
|
||||
}
|
||||
|
||||
SoundStream *OpenALSoundRenderer::OpenStream(const char *filename, int flags, int offset, int length)
|
||||
{
|
||||
std::auto_ptr<OpenALSoundStream> stream(new OpenALSoundStream(this));
|
||||
|
||||
if(offset > 0)
|
||||
{
|
||||
// If there's an offset to the start of the data, separate it into its
|
||||
// own temp file
|
||||
FILE *infile = fopen(filename, "rb");
|
||||
FILE *f = fopen(stream->tmpfile, "wb");
|
||||
if(!infile || !f || fseek(infile, offset, SEEK_SET) != 0)
|
||||
{
|
||||
if(infile) fclose(infile);
|
||||
if(f) fclose(f);
|
||||
return NULL;
|
||||
}
|
||||
BYTE buf[1024];
|
||||
size_t got;
|
||||
do {
|
||||
got = (std::min)(sizeof(buf), (size_t)length);
|
||||
got = fread(buf, 1, got, infile);
|
||||
if(got == 0)
|
||||
break;
|
||||
} while(fwrite(buf, 1, got, f) == got && (length-=got) > 0);
|
||||
fclose(f);
|
||||
fclose(infile);
|
||||
|
||||
filename = stream->tmpfile;
|
||||
}
|
||||
|
||||
bool ok = ((offset == -1) ? stream->Init((const BYTE*)filename, length) :
|
||||
stream->Init(filename));
|
||||
stream->Init(filename, offset, length));
|
||||
if(ok == false)
|
||||
return NULL;
|
||||
|
||||
|
|
|
@ -196,8 +196,9 @@ private:
|
|||
ALuint EnvFilters[2];
|
||||
float LastWaterAbsorb;
|
||||
|
||||
std::vector<OpenALSoundStream*> Streams;
|
||||
std::vector<SoundStream*> Streams;
|
||||
friend class OpenALSoundStream;
|
||||
friend class OpenALCallbackStream;
|
||||
};
|
||||
|
||||
#endif // NO_OPENAL
|
||||
|
|
Loading…
Reference in a new issue