mirror of
https://github.com/OpenMW/openmw.git
synced 2025-02-06 06:15:32 +00:00
first FFmpeg / avcodoc commit (does NOT compile)
git-svn-id: https://openmw.svn.sourceforge.net/svnroot/openmw/trunk@24 ea6a568a-9f4f-0410-981a-c910a81bb256
This commit is contained in:
parent
1974b68dcd
commit
a99139cd2f
9 changed files with 462 additions and 64 deletions
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@ -39,6 +39,7 @@ Dependencies needed to build OpenMW:
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OGRE 1.4.5 (3d engine)
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OIS-1.0.0 (input system)
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OpenAL (3d sound system)
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libavcodec (MP3 library)
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gcc and g++ (C++ compiler)
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GNU make (build tool for C++ files)
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DMD 1.031 (D compiler)
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@ -46,11 +47,11 @@ DMD 1.031 (D compiler)
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The above versions are the ones I have tested recently, but other
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versions might work. OGRE and OIS will require their own set of
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dependencies. I recommend using an automated package tool to install
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as many of these as possible. On ubuntu, try typing:
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versions might work. OGRE, OIS and the other libraries have
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dependencies of their own, so I recommend using an automated package
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tool to install as many of these as possible. On ubuntu, try typing:
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sudo apt-get install libogre-dev libalut0 libois-dev build-essential g++ gdc
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sudo apt-get install libogre-dev libavcodec-dev libois-dev build-essential g++ gdc
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If you want to install Ogre, OpenAL or OIS manually, try:
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OGRE: http://ogre3d.org/
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@ -68,6 +69,9 @@ currently two choices for the D compiler, DMD and GDC. DMD is the
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completely open source frontend to GCC (The GNU compiler.) Both should
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work equally well with OpenMW.
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If you want to install GDC manually, go to
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http://sourceforge.net/projects/dgcc
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If you want to use DMD instead, it can be found at:
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http://digitalmars.com/d/1.0/dmd-linux.html
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23
Makefile
23
Makefile
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@ -11,13 +11,21 @@ NIFFLAGS=
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# Compiler settings for Ogre + OIS. Change as needed.
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OGCC=$(CXX) $(CXXFLAGS) `pkg-config --cflags OGRE OIS openal`
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# Compiler settings for ffmpeg. Change as needed.
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AVGCC=$(CXX) $(CXXFLAGS) `pkg-config --cflags libavcodec libavformat`
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# Ogre C++ files, on the form ogre/cpp_X.cpp. Only the first file is
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# passed to the compiler, the rest are dependencies.
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ogre_cpp=ogre framelistener interface overlay bsaarchive
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# FFmpeg files, in the form sound/cpp_X.cpp. Only the first file is
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# passed to the compiler, the rest are dependencies.
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avcodec_cpp=avcodec
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## The rest of this file is automatic ##
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ogre_cpp_files=$(ogre_cpp:%=ogre/cpp_%.cpp)
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avcodec_cpp_files=$(avcodec_cpp:%=sound/cpp_%.cpp)
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d_files=$(wildcard */*.d) $(wildcard monster/util/*.d)
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d_files_nif=$(wildcard nif/*.d) $(wildcard util/*.d) $(wildcard core/memory.d) $(wildcard monster/util/*.d)
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@ -30,13 +38,16 @@ d_objs_nif=$(d_files_nif:%.d=nifobjs/%.o)
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.PHONY: cpp all clean makedirs
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# By default, make will only build the Ogre C++ sources.
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cpp: cpp_ogre.o
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cpp: cpp_ogre.o cpp_avcodec.o
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all: makedirs openmw esmtool niftool bsatool bored
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cpp_ogre.o: $(ogre_cpp_files)
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$(OGCC) -c $<
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cpp_avcodec.o: $(avcodec_cpp_files)
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$(AVGCC) -c $<
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objs/%.o: %.d makedirs
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$(DMD) -c $< -of$@
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@ -62,11 +73,11 @@ makedirs:
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mkdir -p nifobjs/monster/util
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mkdir -p nifobjs/bsa
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openmw: openmw.d cpp_ogre.o $(d_objs)
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$(DMD) $^ -of$@ -L-lalut -L-lopenal -L-lOgreMain -L-lOIS
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openmw: openmw.d cpp_ogre.o cpp_avcodec.o $(d_objs)
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$(DMD) $^ -of$@ -L-lopenal -L-lOgreMain -L-lOIS -L-lavcodec -L-lavformat
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esmtool: esmtool.d cpp_ogre.o $(d_objs)
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$(DMD) $^ -of$@ -L-lalut -L-lopenal -L-lOgreMain -L-lOIS
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esmtool: esmtool.d cpp_ogre.o cpp_avcodec.o $(d_objs)
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$(DMD) $^ -of$@ -L-lopenal -L-lOgreMain -L-lOIS -L-lavcodec -L-lavformat
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niftool: niftool.d $(d_objs_nif)
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$(DMD) $^ -of$@
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@ -78,6 +89,6 @@ bored: bored.d
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$(DMD) $^
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clean:
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-rm -f cpp_ogre.o
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-rm -f cpp_ogre.o cpp_avcodec.o bored.o bsafile.o bsatool.o esmtool.o niftool.o openmw.o
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-rm -f openmw esmtool niftool bsatool bored
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-rm -rf objs/ nifobjs/ dsss_objs/
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@ -4,6 +4,6 @@
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make || exit 1
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gdc -Wall -g -fversion=Posix -o openmw openmw.d core/*.d ogre/*.d nif/*.d util/*.d bsa/*.d monster/util/*.d input/*.d sound/*.d scene/*.d esm/*.d cpp_ogre.o -lalut -lopenal -lm -lOgreMain -lOIS -lstdc++
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gdc -Wall -g -fversion=Posix -o openmw openmw.d core/*.d ogre/*.d nif/*.d util/*.d bsa/*.d monster/util/*.d input/*.d sound/*.d scene/*.d esm/*.d cpp_ogre.o cpp_avcodec.o -lopenal -lm -lOgreMain -lOIS -lavcodec -lavformat -lstdc++
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gdc -Wall -g -fversion=Posix -o esmtool esmtool.d core/*.d ogre/*.d nif/*.d util/*.d bsa/*.d monster/util/*.d input/*.d sound/*.d scene/*.d esm/*.d cpp_ogre.o -lalut -lopenal -lm -lOgreMain -lOIS -lstdc++
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gdc -Wall -g -fversion=Posix -o esmtool esmtool.d core/*.d ogre/*.d nif/*.d util/*.d bsa/*.d monster/util/*.d input/*.d sound/*.d scene/*.d esm/*.d cpp_ogre.o cpp_avcodec.o -lopenal -lm -lOgreMain -lOIS -lavcodec -lavformat -lstdc++
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@ -4,7 +4,7 @@
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[openmw.d]
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# Add libraries and the two C++ object files
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buildflags = -llOgreMain -llalut -llopenal -llOIS cpp_ogre.o
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buildflags = -llOgreMain -llopenal -llOIS -llavcodec -llavformat cpp_ogre.o cpp_avcodec.o
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# Make sure the C++ object files are built first
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version(Windows) {
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@ -34,7 +34,7 @@ prebuild += dsss clean niftool
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[esmtool.d]
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# Because of interdepencies between the ESM code and the resource
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# manager, we have to include all the C++ stuff.
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buildflags = -llOgreMain -llalut -llopenal -llOIS cpp_ogre.o
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buildflags = -llOgreMain -llopenal -llOIS -llavcodec -llavformat cpp_ogre.o cpp_avcodec.o
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@ -32,13 +32,6 @@ import sound.alc;
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import std.stdio;
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import std.string;
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ALCdevice *Device = null;
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ALCcontext *Context = null;
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// Temporarilly use ALUT for data loading until something better is picked
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extern (C) ALboolean alutInitWithoutContext(int *argc, char **argv);
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extern (C) ALboolean alutExit();
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class SoundException : Exception
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{
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this(char[] caller, char[] msg) { super(caller ~ " SoundException: " ~ msg); }
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@ -47,6 +40,9 @@ class SoundException : Exception
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MusicManager jukebox;
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MusicManager battleMusic;
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ALCdevice *Device = null;
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ALCcontext *Context = null;
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void initializeSound()
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{
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Device = alcOpenDevice(null);
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@ -57,7 +53,6 @@ void initializeSound()
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"Failed to initialize music device");
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alcMakeContextCurrent(Context);
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alutInitWithoutContext(null, null);
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// Gross HACK: We should use the default model (inverse distance clamped).
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// But without a proper rolloff factor, distance attenuation is completely
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@ -73,8 +68,6 @@ void shutdownSound()
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jukebox.disableMusic();
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battleMusic.disableMusic();
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alutExit();
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alcMakeContextCurrent(null);
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if(Context) alcDestroyContext(Context);
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Context = null;
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40
sound/avcodec.d
Normal file
40
sound/avcodec.d
Normal file
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@ -0,0 +1,40 @@
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extern (C):
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// A unique handle that represents an AV file
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typedef void* AVFile;
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// A unique handle representing an audio stream
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typedef void* AVAudio;
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// In case we ever decide to implement more codec backends, here's what
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// these functions need to do...
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// Open the named file, and return a unique handle representing it.
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// Returns NULL on error
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AVFile cpp_openAVFile(char *fname);
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// Close the file handle, invalidating all streams taken from it
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void cpp_closeAVFile(AVFile file);
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// Get a unique handle to an audio stream in the file. The given number
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// is for files that can contain multiple audio streams (generally you
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// would pass 0, for the first audio stream)
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void *cpp_getAVAudioStream(AVFile file, int streamnum);
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// Get audio info representing the current stream. Returns 0 for success
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// (not likely to fail)
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int cpp_getAVAudioInfo(AVAudio stream, int *rate, int *channels, int *bits);
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// Decode the next bit of data for the given audio stream. The function
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// must provide no less than the requested number of bytes, except for
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// end-of-stream conditions, and is responsible for buffering data. For
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// files with multiple streams, it must take care to preserve data for
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// any stream that has had a stream handle returned.
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// eg. if a file has one video stream and 2 audio streams and the app
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// gets a handle to the video stream and one audio stream, it must
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// not destroy video data for subsequent calls to cpp_getAVVideoData if
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// it has to read over it while decoding the audio stream. The other
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// audio stream's data, however, may be discarded.
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// Returns the number of bytes written to the buffer, which will be no
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// more than the provided length.
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int cpp_getAVAudioData(AVAudio stream, void *data, int length);
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209
sound/cpp_avcodec.cpp
Normal file
209
sound/cpp_avcodec.cpp
Normal file
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@ -0,0 +1,209 @@
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/*
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OpenMW - The completely unofficial reimplementation of Morrowind
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Copyright (C) 2008 Nicolay Korslund
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Email: < korslund@gmail.com >
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WWW: http://openmw.snaptoad.com/
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This file (cpp_avcodec.cpp) is part of the OpenMW package.
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OpenMW is distributed as free software: you can redistribute it
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and/or modify it under the terms of the GNU General Public License
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version 3, as published by the Free Software Foundation.
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This program is distributed in the hope that it will be useful, but
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WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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General Public License for more details.
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You should have received a copy of the GNU General Public License
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version 3 along with this program. If not, see
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http://www.gnu.org/licenses/ .
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*/
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#include <stdio.h>
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extern "C" { // the headers don't do this..
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#include <ffmpeg/avcodec.h>
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#include <ffmpeg/avformat.h>
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}
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#include <vector>
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using std::vector;
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struct MyFile {
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AVFormatContext *FmtCtx;
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struct MyStream {
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MyFile *parent;
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AVCodecContext *CodecCtx;
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int StreamNum;
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vector<uint8_t> Data;
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vector<char> DecodedData;
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};
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vector<MyStream> Streams;
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};
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// TODO:
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// extern "C" MyFile::MyStream *cpp_getAVVideoStream(MyFile *file, int streamnum);
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// extern "C" int cpp_getAVVideoInfo(MyFile::MyStream *stream, float *fps, int *width, int * height);
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// extern "C" int cpp_getAVVideoData(MyFile::MyStream *stream, char *data, int length);
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extern "C" MyFile *cpp_openAVFile(char *fname)
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{
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static bool done = false;
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if(!done) { av_register_all();
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av_log_set_level(AV_LOG_ERROR);}
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done = true;
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MyFile *file = new MyFile;
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if(av_open_input_file(&file->FmtCtx, fname, NULL, 0, NULL) == 0)
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{
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if(av_find_stream_info(file->FmtCtx) >= 0)
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return file;
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av_close_input_file(file->FmtCtx);
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}
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delete file;
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return NULL;
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}
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extern "C" void cpp_closeAVFile(MyFile *file)
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{
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if(!file) return;
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for(size_t i = 0;i < file->Streams.size();i++)
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{
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avcodec_close(file->Streams[i].CodecCtx);
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file->Streams[i].Data.clear();
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file->Streams[i].DecodedData.clear();
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}
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file->Streams.clear();
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av_close_input_file(file->FmtCtx);
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delete file;
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}
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extern "C" MyFile::MyStream *cpp_getAVAudioStream(MyFile *file, int streamnum)
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{
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if(!file) return NULL;
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for(unsigned int i = 0;i < file->FmtCtx->nb_streams;i++)
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{
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if(file->FmtCtx->streams[i]->codec->codec_type != CODEC_TYPE_AUDIO)
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continue;
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if(streamnum == 0)
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{
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MyFile::MyStream stream;
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stream.parent = file;
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stream.CodecCtx = file->FmtCtx->streams[i]->codec;
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stream.StreamNum = i;
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AVCodec *codec = avcodec_find_decoder(stream.CodecCtx->codec_id);
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if(!codec) return NULL;
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if(avcodec_open(stream.CodecCtx, codec) < 0)
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return NULL;
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file->Streams.push_back(stream);
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return &file->Streams[file->Streams.size()-1];
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}
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streamnum--;
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||||
}
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return NULL;
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}
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extern "C" int cpp_getAVAudioInfo(MyFile::MyStream *stream,
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int *rate, int *channels, int *bits)
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{
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if(!stream) return 1;
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if(rate) *rate = stream->CodecCtx->sample_rate;
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if(channels) *channels = stream->CodecCtx->channels;
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if(bits) *bits = 16;
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||||
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return 0;
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||||
}
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||||
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||||
static int getNextPacket(MyFile *file)
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||||
{
|
||||
AVPacket packet;
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while(av_read_frame(file->FmtCtx, &packet) >= 0)
|
||||
{
|
||||
for(vector<MyFile::MyStream>::iterator i = file->Streams.begin();
|
||||
i != file->Streams.end();i++)
|
||||
{
|
||||
if(i->StreamNum == packet.stream_index)
|
||||
{
|
||||
size_t idx = i->Data.size();
|
||||
i->Data.resize(idx + packet.size);
|
||||
memcpy(&i->Data[idx], packet.data, packet.size);
|
||||
av_free_packet(&packet);
|
||||
return 0;
|
||||
}
|
||||
}
|
||||
av_free_packet(&packet);
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||||
}
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||||
|
||||
return 1;
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||||
}
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||||
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||||
extern "C" int cpp_getAVAudioData(MyFile::MyStream *stream, char *data, int length)
|
||||
{
|
||||
if(!stream) return 0;
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||||
|
||||
int dec = 0;
|
||||
while(dec < length)
|
||||
{
|
||||
if(stream->DecodedData.size() == 0)
|
||||
{
|
||||
while(stream->Data.size() == 0)
|
||||
{
|
||||
if(getNextPacket(stream->parent) != 0)
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||||
break;
|
||||
}
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||||
|
||||
int insize = stream->Data.size();
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||||
if(insize == 0)
|
||||
break;
|
||||
|
||||
// Temporarilly add padding to the input data since some
|
||||
// codecs read in larger chunks and may accidently read
|
||||
// past the end of the allocated buffer
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||||
stream->Data.resize(insize + FF_INPUT_BUFFER_PADDING_SIZE);
|
||||
stream->DecodedData.resize(AVCODEC_MAX_AUDIO_FRAME_SIZE);
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||||
|
||||
int16_t *ptr = (int16_t*)&stream->DecodedData[0];
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||||
int size = stream->DecodedData.size();
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||||
int len = avcodec_decode_audio2(stream->CodecCtx, ptr, &size,
|
||||
&stream->Data[0], insize);
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||||
if(len < 0)
|
||||
{
|
||||
stream->Data.resize(insize);
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||||
break;
|
||||
}
|
||||
|
||||
int datarem = insize-len;
|
||||
if(datarem)
|
||||
memmove(&stream->Data[0], &stream->Data[len], datarem);
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||||
stream->Data.resize(datarem);
|
||||
|
||||
stream->DecodedData.resize(size);
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||||
if(stream->DecodedData.size() == 0)
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||||
break;
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||||
}
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||||
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||||
size_t rem = length-dec;
|
||||
if(rem > stream->DecodedData.size())
|
||||
rem = stream->DecodedData.size();
|
||||
|
||||
memcpy(data, &stream->DecodedData[0], rem);
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||||
data += rem;
|
||||
dec += rem;
|
||||
if(rem < stream->DecodedData.size())
|
||||
memmove(&stream->DecodedData[0], &stream->DecodedData[rem],
|
||||
stream->DecodedData.size() - rem);
|
||||
stream->DecodedData.resize(stream->DecodedData.size()-rem);
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||||
}
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||||
return dec;
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||||
}
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141
sound/music.d
141
sound/music.d
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@ -23,6 +23,7 @@
|
|||
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||||
module sound.music;
|
||||
|
||||
import sound.avcodec;
|
||||
import sound.audio;
|
||||
import sound.al;
|
||||
|
||||
|
@ -32,10 +33,6 @@ import std.string;
|
|||
import core.config;
|
||||
import core.resource;
|
||||
|
||||
extern (C) ALuint alutCreateBufferFromFile(char *filename);
|
||||
extern (C) ALenum alutGetError();
|
||||
extern (C) ALchar *alutGetErrorString(ALenum err);
|
||||
|
||||
// Simple music player, has a playlist and can pause/resume music.
|
||||
struct MusicManager
|
||||
{
|
||||
|
@ -45,6 +42,9 @@ struct MusicManager
|
|||
const float fadeInRate = 0.10;
|
||||
const float fadeOutRate = 0.35;
|
||||
|
||||
// Maximum buffer length, divided up among OpenAL buffers
|
||||
const uint bufLength = 128*1024;
|
||||
|
||||
// Volume
|
||||
ALfloat volume, maxVolume;
|
||||
|
||||
|
@ -61,9 +61,14 @@ struct MusicManager
|
|||
|
||||
bool musicOn;
|
||||
ALuint sID;
|
||||
ALuint bIDs[1];
|
||||
ALuint bIDs[4];
|
||||
|
||||
ubyte[] readData;
|
||||
ALenum bufFormat;
|
||||
ALint bufRate;
|
||||
|
||||
AVFile fileHandle;
|
||||
AVAudio audioHandle;
|
||||
ubyte[] outData;
|
||||
|
||||
// Which direction are we currently fading, if any
|
||||
enum Fade { None = 0, In, Out }
|
||||
|
@ -77,6 +82,8 @@ struct MusicManager
|
|||
this.name = name;
|
||||
sID = 0;
|
||||
foreach(ref b; bIDs) b = 0;
|
||||
outData.length = bufLength / bIDs.length;
|
||||
fileHandle = null;
|
||||
musicOn = false;
|
||||
updateVolume();
|
||||
}
|
||||
|
@ -156,6 +163,10 @@ struct MusicManager
|
|||
foreach(ref b; bIDs) b = 0;
|
||||
checkALError();
|
||||
|
||||
if(fileHandle) cpp_closeAVFile(fileHandle);
|
||||
fileHandle = null;
|
||||
audioHandle = null;
|
||||
|
||||
// End of list? Randomize and start over
|
||||
if(index == playlist.length)
|
||||
{
|
||||
|
@ -163,37 +174,97 @@ struct MusicManager
|
|||
index = 0;
|
||||
}
|
||||
|
||||
readData.length = 128*1024 / bIDs.length;
|
||||
|
||||
// FIXME: Should load up and queue 3 or 4 buffers here instead of trying to
|
||||
// load it all into one (when we switch away from ALUT).
|
||||
char *fname = toStringz(playlist[index]);
|
||||
bIDs[0] = alutCreateBufferFromFile(fname);
|
||||
if(!bIDs[0])
|
||||
alGenBuffers(bIDs.length, bIDs.ptr);
|
||||
if(checkALError() != AL_NO_ERROR)
|
||||
{
|
||||
writefln("Unable to load music track %s: %s", playlist[index],
|
||||
toString(alutGetErrorString(alutGetError())));
|
||||
writefln("Unable to create %d buffers", bIDs.length);
|
||||
alDeleteSources(1, &sID);
|
||||
checkALError();
|
||||
sID = 0;
|
||||
index++;
|
||||
return;
|
||||
}
|
||||
|
||||
alSourcei(sID, AL_BUFFER, bIDs[0]);
|
||||
fileHandle = cpp_openAVFile(toStringz(playlist[index]));
|
||||
if(!fileHandle)
|
||||
{
|
||||
writefln("Unable to open %s", playlist[index]);
|
||||
goto errclose;
|
||||
}
|
||||
|
||||
audioHandle = cpp_getAVAudioStream(fileHandle, 0);
|
||||
if(!audioHandle)
|
||||
{
|
||||
writefln("Unable to load music track %s", playlist[index]);
|
||||
goto errclose;
|
||||
}
|
||||
|
||||
int ch, bits, rate;
|
||||
if(cpp_getAVAudioInfo(audioHandle, &rate, &ch, &bits) != 0)
|
||||
{
|
||||
writefln("Unable to get info for music track %s", playlist[index]);
|
||||
goto errclose;
|
||||
}
|
||||
|
||||
bufRate = rate;
|
||||
bufFormat = 0;
|
||||
if(bits == 8)
|
||||
{
|
||||
if(ch == 1) bufFormat = AL_FORMAT_MONO8;
|
||||
if(ch == 2) bufFormat = AL_FORMAT_STEREO8;
|
||||
if(ch == 4) bufFormat = alGetEnumValue("AL_FORMAT_QUAD8");
|
||||
}
|
||||
if(bits == 16)
|
||||
{
|
||||
if(ch == 1) bufFormat = AL_FORMAT_MONO16;
|
||||
if(ch == 2) bufFormat = AL_FORMAT_STEREO16;
|
||||
if(ch == 4) bufFormat = alGetEnumValue("AL_FORMAT_QUAD16");
|
||||
}
|
||||
|
||||
if(bufFormat == 0)
|
||||
{
|
||||
writefln("Unhandled format (%d channels, %d bits) for music track %s", ch, bits, playlist[index]);
|
||||
goto errclose;
|
||||
}
|
||||
|
||||
foreach(int i, ref b; bIDs)
|
||||
{
|
||||
int length = cpp_getAVAudioData(audioHandle, outData.ptr, outData.length);
|
||||
if(length) alBufferData(b, bufFormat, outData.ptr, length, bufRate);
|
||||
if(length == 0 || checkALError() != AL_NO_ERROR)
|
||||
{
|
||||
if(i == 0)
|
||||
{
|
||||
writefln("No audio data in music track %s", playlist[index]);
|
||||
goto errclose;
|
||||
}
|
||||
alDeleteBuffers(bIDs.length-i, bIDs.ptr+i);
|
||||
checkALError();
|
||||
bIDs[i..$] = 0;
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
alSourceQueueBuffers(sID, bIDs.length, bIDs.ptr);
|
||||
alSourcePlay(sID);
|
||||
if(checkALError() != AL_NO_ERROR)
|
||||
{
|
||||
writefln("Unable to start music track %s", playlist[index]);
|
||||
alSourceStop(sID);
|
||||
alDeleteSources(1, &sID);
|
||||
alDeleteBuffers(bIDs.length, bIDs.ptr);
|
||||
checkALError();
|
||||
sID = 0;
|
||||
foreach(ref b; bIDs) b = 0;
|
||||
goto errclose;
|
||||
}
|
||||
|
||||
index++;
|
||||
return;
|
||||
errclose:
|
||||
if(fileHandle) cpp_closeAVFile(fileHandle);
|
||||
fileHandle = null;
|
||||
audioHandle = null;
|
||||
alSourceStop(sID);
|
||||
alDeleteSources(1, &sID);
|
||||
alDeleteBuffers(bIDs.length, bIDs.ptr);
|
||||
checkALError();
|
||||
sID = 0;
|
||||
foreach(ref b; bIDs) b = 0;
|
||||
index++;
|
||||
}
|
||||
|
||||
// Start playing the jukebox
|
||||
|
@ -210,15 +281,22 @@ struct MusicManager
|
|||
// Disable music
|
||||
void disableMusic()
|
||||
{
|
||||
if(fileHandle) cpp_closeAVFile(fileHandle);
|
||||
fileHandle = null;
|
||||
audioHandle = null;
|
||||
|
||||
if(sID)
|
||||
{
|
||||
alSourceStop(sID);
|
||||
alDeleteSources(1, &sID);
|
||||
alDeleteBuffers(bIDs.length, bIDs.ptr);
|
||||
checkALError();
|
||||
sID = 0;
|
||||
foreach(ref b; bIDs) b = 0;
|
||||
}
|
||||
|
||||
alDeleteBuffers(bIDs.length, bIDs.ptr);
|
||||
checkALError();
|
||||
foreach(ref b; bIDs) b = 0;
|
||||
|
||||
musicOn = false;
|
||||
}
|
||||
|
||||
|
@ -245,14 +323,14 @@ struct MusicManager
|
|||
private bool isPlaying()
|
||||
{
|
||||
if(!sID) return false;
|
||||
/* Use this when we can do streaming..
|
||||
|
||||
ALint count;
|
||||
alGetSourcei(sID, AL_BUFFERS_PROCESSED, &count);
|
||||
if(checkALError() != AL_NO_ERROR) return false;
|
||||
|
||||
for(int i = 0;i < count;i++)
|
||||
{
|
||||
int length = GetData(readData.ptr, readData.length);
|
||||
int length = cpp_getAVAudioData(audioHandle, outData.ptr, outData.length);
|
||||
if(length <= 0)
|
||||
{
|
||||
if(i == 0)
|
||||
|
@ -269,7 +347,7 @@ struct MusicManager
|
|||
alSourceUnqueueBuffers(sID, 1, &bid);
|
||||
if(checkALError() == AL_NO_ERROR)
|
||||
{
|
||||
alBufferData(bid, dataFreq, dataFormat, length, readData.ptr);
|
||||
alBufferData(bid, bufFormat, outData.ptr, length, bufRate);
|
||||
alSourceQueueBuffers(sID, 1, &bid);
|
||||
checkALError();
|
||||
}
|
||||
|
@ -279,13 +357,6 @@ struct MusicManager
|
|||
alGetSourcei(sID, AL_SOURCE_STATE, &state);
|
||||
if(state != AL_PLAYING) alSourcePlay(sID);
|
||||
return (checkALError() == AL_NO_ERROR);
|
||||
*/
|
||||
ALint state;
|
||||
alGetSourcei(sID, AL_SOURCE_STATE, &state);
|
||||
if(checkALError() != AL_NO_ERROR || state == AL_STOPPED) return false;
|
||||
|
||||
if(state != AL_PLAYING) alSourcePlay(sID);
|
||||
return (checkALError() == AL_NO_ERROR);
|
||||
}
|
||||
|
||||
// Check if the music has died. This function is also used for fading.
|
||||
|
|
80
sound/sfx.d
80
sound/sfx.d
|
@ -23,6 +23,7 @@
|
|||
|
||||
module sound.sfx;
|
||||
|
||||
import sound.avcodec;
|
||||
import sound.audio;
|
||||
import sound.al;
|
||||
|
||||
|
@ -30,8 +31,7 @@ import core.config;
|
|||
import core.resource;
|
||||
|
||||
import std.string;
|
||||
|
||||
extern (C) ALuint alutCreateBufferFromFile(char *filename);
|
||||
import std.stdio;
|
||||
|
||||
// Handle for a sound resource. This struct represents one sound
|
||||
// effect file (not a music file, those are handled differently
|
||||
|
@ -57,11 +57,81 @@ struct SoundFile
|
|||
{
|
||||
name = file;
|
||||
|
||||
loaded = true;
|
||||
loaded = false;
|
||||
refs = 0;
|
||||
bID = 0;
|
||||
|
||||
bID = alutCreateBufferFromFile(toStringz(file));
|
||||
if(!bID) fail("Failed to open sound file " ~ file);
|
||||
ubyte[] outData;
|
||||
AVFile fileHandle = cpp_openAVFile(toStringz(file));
|
||||
AVAudio audioHandle = cpp_getAVAudioStream(fileHandle, 0);
|
||||
|
||||
if(!fileHandle)
|
||||
{
|
||||
writefln("Unable to open %s", file);
|
||||
goto errclose;
|
||||
}
|
||||
if(!audioHandle)
|
||||
{
|
||||
writefln("Unable to load sound %s", file);
|
||||
goto errclose;
|
||||
}
|
||||
|
||||
int ch, bits, rate;
|
||||
if(cpp_getAVAudioInfo(audioHandle, &rate, &ch, &bits) != 0)
|
||||
{
|
||||
writefln("Unable to get info for sound %s", file);
|
||||
goto errclose;
|
||||
}
|
||||
|
||||
int fmt = 0;
|
||||
if(bits == 8)
|
||||
{
|
||||
if(ch == 1) fmt = AL_FORMAT_MONO8;
|
||||
if(ch == 2) fmt = AL_FORMAT_STEREO8;
|
||||
if(ch == 4) fmt = alGetEnumValue("AL_FORMAT_QUAD8");
|
||||
}
|
||||
if(bits == 16)
|
||||
{
|
||||
if(ch == 1) fmt = AL_FORMAT_MONO16;
|
||||
if(ch == 2) fmt = AL_FORMAT_STEREO16;
|
||||
if(ch == 4) fmt = alGetEnumValue("AL_FORMAT_QUAD16");
|
||||
}
|
||||
|
||||
if(fmt == 0)
|
||||
{
|
||||
writefln("Unhandled format (%d channels, %d bits) for sound %s", ch, bits, file);
|
||||
goto errclose;
|
||||
}
|
||||
|
||||
int total = 0;
|
||||
do
|
||||
{
|
||||
// Grow by an arbitrary amount. Should be big enough to get the
|
||||
// whole sound in one or two iterations, but not allocate too much
|
||||
// memory in case its short
|
||||
outData.length = outData.length+8192;
|
||||
int length = cpp_getAVAudioData(audioHandle, outData.ptr+total, outData.length-total);
|
||||
total += length;
|
||||
}
|
||||
while(total == outData.length);
|
||||
|
||||
if(total)
|
||||
{
|
||||
alGenBuffers(1, &bID);
|
||||
alBufferData(bID, fmt, outData.ptr, total, rate);
|
||||
if(checkALError() != AL_NO_ERROR)
|
||||
{
|
||||
writefln("Unable to load sound %s", file);
|
||||
alDeleteBuffers(1, &bID);
|
||||
bID = 0;
|
||||
}
|
||||
else loaded = true;
|
||||
}
|
||||
|
||||
errclose:
|
||||
if(fileHandle) cpp_closeAVFile(fileHandle);
|
||||
fileHandle = null;
|
||||
audioHandle = null;
|
||||
}
|
||||
|
||||
// Get an instance of this resource.
|
||||
|
|
Loading…
Reference in a new issue