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Use SDL_sound when available to decode files

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Chris Robinson 2013-06-30 07:44:19 -07:00
parent 99209c4a2a
commit 69af01d629
6 changed files with 2744 additions and 1578 deletions
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382
FindSDL_sound.cmake Normal file
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@ -0,0 +1,382 @@
# - Locates the SDL_sound library
#
# This module depends on SDL being found and
# must be called AFTER FindSDL.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 to override this selectionor 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(SDL_FOUND AND 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 SDL_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 FindSDL.cmake module, but that's kind of painful.
# The solution would be to try something like:
# set(SDL_TRY_COMPILE_LIBRARY_LIST "${SDL_TRY_COMPILE_LIBRARY_LIST}\;${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}
# ${PROJECT_SOURCE_DIR}/DetermineSoundLibs.c
# CMAKE_FLAGS
# -DINCLUDE_DIRECTORIES:STRING=${SDL_INCLUDE_DIR}\;${SDL_SOUND_INCLUDE_DIR}
# -DLINK_LIBRARIES:STRING=${SDL_SOUND_LIBRARY}\;${SDL_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} ${SDL_LIBRARY})
# causes problems when SDL_LIBRARY looks like
# /Library/Frameworks/SDL.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_TRY_LIBS)
foreach(lib ${SDL_SOUND_LIBRARY} ${SDL_LIBRARY})
set(TMP_TRY_LIBS "${TMP_TRY_LIBS} \"${lib}\"")
endforeach()
# 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 SDL_LIBRARY string after the "-framework".
# But if I quote the stuff in INCLUDE_DIRECTORIES, it doesn't work.
file(WRITE ${PROJECT_BINARY_DIR}/CMakeTmp/CMakeLists.txt
"cmake_minimum_required(VERSION 2.8)
project(DetermineSoundLibs C)
include_directories(${SDL_INCLUDE_DIR} ${SDL_SOUND_INCLUDE_DIR})
add_executable(DetermineSoundLibs DetermineSoundLibs.c)
target_link_libraries(DetermineSoundLibs ${TMP_TRY_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.
set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARY})
# Find MikMod
if("${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} ${MIKMOD_LIBRARY})
endif(MIKMOD_LIBRARY)
endif("${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} ${MODPLUG_LIBRARY})
endif()
endif()
# Find Ogg and Vorbis
if("${MY_OUTPUT}" MATCHES "ov_")
find_library(VORBIS_LIBRARY
NAMES vorbis Vorbis VORBIS
PATHS
ENV VORBISDIR
ENV OGGDIR
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})
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} ${OGG_LIBRARY})
endif()
endif()
# Find SMPEG
if("${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} ${SMPEG_LIBRARY})
endif()
endif()
# Find FLAC
if("${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} ${FLAC_LIBRARY})
endif()
endif()
# Hmmm...Speex seems to depend on Ogg. This might be a problem if
# the TRY_COMPILE attempt gets blocked at SPEEX before it can pull
# in the Ogg symbols. I'm not sure if I should duplicate the ogg stuff
# above for here or if two ogg entries will screw up things.
if("${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})
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})
endif()
endif()
endif()
set(SDL_SOUND_LIBRARIES ${SDL_SOUND_EXTRAS} ${SDL_SOUND_LIBRARIES_TMP} CACHE INTERNAL "SDL_sound and dependent libraries")
else()
set(SDL_SOUND_LIBRARIES ${SDL_SOUND_EXTRAS} ${SDL_SOUND_LIBRARY} CACHE INTERNAL "SDL_sound and dependent libraries")
endif()
endif()
if(SDL_SOUND_INCLUDE_DIR AND EXISTS "${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_LIBRARY SDL_SOUND_INCLUDE_DIR
VERSION_VAR SDL_SOUND_VERSION_STRING)

13
src/CMakeLists.txt
View file

@ -202,11 +202,22 @@ else( WIN32 )
endif( WIN32 )
set( OAL_SOURCES sound/oalsound.cpp )
if( NOT NO_OPENAL )
find_package( OpenAL )
if( OPENAL_FOUND )
include_directories( ${OPENAL_INCLUDE_DIR} )
set( ZDOOM_LIBS ${OPENAL_LIBRARY} ${ZDOOM_LIBS} )
find_package( SDL )
if( SDL_FOUND )
find_package( SDL_sound )
if( SDL_SOUND_FOUND )
set_source_files_properties( sound/oalsound.cpp PROPERTIES COMPILE_FLAGS "-DWITH_SDL_SOUND=1" )
set( OAL_SOURCES ${OAL_SOURCES} sound/oalsdlsound.cpp )
include_directories( ${SDL_SOUND_INCLUDE_DIR} )
set( ZDOOM_LIBS ${SDL_SOUND_LIBRARIES} ${ZDOOM_LIBS} )
endif( SDL_SOUND_FOUND )
endif( SDL_FOUND )
else( OPENAL_FOUND )
set( NO_OPENAL ON )
endif( OPENAL_FOUND )
@ -900,7 +911,7 @@ add_executable( zdoom WIN32
sound/music_softsynth_mididevice.cpp
sound/music_timidity_mididevice.cpp
sound/music_win_mididevice.cpp
sound/oalsound.cpp
${OAL_SOURCES}
sound/music_pseudo_mididevice.cpp
textures/animations.cpp
textures/anim_switches.cpp

494
src/sound/oalsdlsound.cpp Normal file
View file

@ -0,0 +1,494 @@
/*
** oalsdlsound.cpp
** Interface for SDL_sound; uses OpenAL
**
**---------------------------------------------------------------------------
** Copyright 2008-2010 Chris Robinson
** 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
View 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 */

502
src/sound/oalsound.cpp
View file

@ -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;

3
src/sound/oalsound.h
View file

@ -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