mirror of
https://github.com/AquariaOSE/Aquaria.git
synced 2024-11-15 14:09:06 +00:00
1815 lines
52 KiB
C++
1815 lines
52 KiB
C++
/*
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Copyright (C) 2007, 2010 - Bit-Blot
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This file is part of Aquaria.
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Aquaria is free software; you can redistribute it and/or
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modify it under the terms of the GNU General Public License
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as published by the Free Software Foundation; either version 2
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of the License, or (at your option) any later version.
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This program is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
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See the GNU 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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along with this program; if not, write to the Free Software
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Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
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*/
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// This file implements just enough of the FMOD library with OpenAL to suit
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// the needs of the existing game code without having to actually ship FMOD.
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <assert.h>
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#ifdef BBGE_BUILD_UNIX
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#include <signal.h>
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#endif
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#include "Base.h"
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#include "Core.h"
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#include "FmodOpenALBridge.h"
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#include "al.h"
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#include "alc.h"
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#include "ogg/ogg.h"
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#include "vorbis/vorbisfile.h"
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#include "MT.h"
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#ifndef _DEBUG
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//#define _DEBUG 1
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#endif
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#undef min
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#undef max
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// HACK: global because OpenAL has only one listener anyway
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static FMOD_VECTOR s_listenerPos;
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///////////////////////////////////////////////////////////////////////////
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// Decoder implementation for streamed Ogg Vorbis audio.
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class OggDecoder {
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public:
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// Create a decoder that streams from a file.
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OggDecoder(VFILE *fp);
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// Create a decoder that streams from a memory buffer.
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OggDecoder(const void *data, long data_size);
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~OggDecoder();
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// Prepare playing on the given channel.
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bool preStart(ALuint source);
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// Decodes the first few buffers, starts the actual playback and detaches the decoder
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// from the main thread, with optional looping.
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void start(bool loop);
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// Decode audio into any free buffers. Must be called periodically
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// on systems without threads; may be called without harm on systems
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// with threads (the function does nothing in that case).
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void update();
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// Terminate playback.
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void stop();
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// Return the current playback position in seconds.
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double position();
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// Memory buffer I/O callback functions for libvorbisfile.
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static size_t mem_read(void *ptr, size_t size, size_t nmemb, void *datasource);
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static int mem_seek(void *datasource, ogg_int64_t offset, int whence);
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static long mem_tell(void *datasource);
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static void startDecoderThread();
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static void stopDecoderThread();
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int getNumChannels() const { return channels; }
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void setForceMono(bool mono) { forcemono = mono; }
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private:
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void _stop();
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// Decoding loop, run in a separate thread (if threads are available).
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static void decode_loop(OggDecoder *this_);
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// Decode and queue PCM data for one buffer; does nothing if the end
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// of the stream has already been reached or an unrecoverable error
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// has occurred during decoding. If looping, the audio will instead
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// restart at the beginning of the stream after reaching the end,
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// but will still stop on an unrecoverable error.
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void queue(ALuint buffer);
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static const int NUM_BUFFERS = 8;
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static const int BUFFER_LENGTH = 4096; // In samples (arbitrary)
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char pcm_buffer[BUFFER_LENGTH * 4]; // Temporary buffer for decoding
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ALuint buffers[NUM_BUFFERS];
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ALuint source;
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// Data source. If fp != NULL, the source is that file; otherwise, the
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// source is the buffer pointed to by "data" with size "data_size" bytes.
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VFILE *fp;
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const char *data;
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long data_size;
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long data_pos; // Current read position for memory buffers
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OggVorbis_File vf;
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ALenum format;
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int channels;
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int freq;
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bool forcemono;
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bool thread; // true if played by background thread
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bool playing;
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bool loop;
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bool eof; // End of file _or_ unrecoverable error encountered
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bool stopped; // true if enqueued deletion
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unsigned int samples_done; // Number of samples played and dequeued
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#ifdef BBGE_BUILD_SDL
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static SDL_Thread *decoderThread;
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static LockedQueue<OggDecoder*> decoderQ;
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static volatile bool stop_thread;
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static std::list<OggDecoder*> decoderList; // used by decoder thread only
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#else
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#warning Threads not supported, music may cut out on area changes!
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// ... because the stream runs out of decoded data while the area is
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// still loading, so OpenAL aborts playback.
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#endif
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static void detachDecoder(OggDecoder *);
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};
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// File I/O callback set (OV_CALLBACKS_NOCLOSE from libvorbis 1.2.0).
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// It might be better to just update libogg/libvorbis to the current
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// versions so we don't have to worry about identifier collisions --
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// we can then drop all this and use OV_CALLBACKS_NOCLOSE in the
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// ov_open_callbacks() call. Note that we rename the fseek() wrapper
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// to avoid an identifier collision when building with more recent
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// versions of libvorbis.
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static int BBGE_ov_header_fseek_wrap(VFILE *f,ogg_int64_t off,int whence){
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if(f==NULL)return(-1);
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return vfseek(f,(long int)off,whence); // no ogg file is larger than 4 GB, int-cast should be ok
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}
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static int noclose(FILE *f) {return 0;}
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static const ov_callbacks local_OV_CALLBACKS_NOCLOSE = {
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(size_t (*)(void *, size_t, size_t, void *)) vfread,
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(int (*)(void *, ogg_int64_t, int)) BBGE_ov_header_fseek_wrap,
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(int (*)(void *)) noclose, // NULL doesn't work in libvorbis-1.1.2
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(long (*)(void *)) vftell
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};
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// Memory I/O callback set.
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static const ov_callbacks ogg_memory_callbacks = {
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OggDecoder::mem_read,
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OggDecoder::mem_seek,
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(int (*)(void *))noclose,
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OggDecoder::mem_tell
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};
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#ifdef BBGE_BUILD_SDL
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SDL_Thread *OggDecoder::decoderThread = NULL;
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LockedQueue<OggDecoder*> OggDecoder::decoderQ;
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volatile bool OggDecoder::stop_thread;
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std::list<OggDecoder*> OggDecoder::decoderList;
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#endif
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void OggDecoder::startDecoderThread()
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{
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#ifdef BBGE_BUILD_SDL
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stop_thread = false;
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#ifdef BBGE_BUILD_SDL2
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decoderThread = SDL_CreateThread((int (*)(void *))decode_loop, "OggDecoder", NULL);
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#else
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decoderThread = SDL_CreateThread((int (*)(void *))decode_loop, NULL);
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#endif
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if (!decoderThread)
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{
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debugLog("Failed to create Ogg Vorbis decode thread: "
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+ std::string(SDL_GetError()));
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}
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#endif
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}
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void OggDecoder::stopDecoderThread()
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{
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#ifdef BBGE_BUILD_SDL
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if (decoderThread)
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{
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stop_thread = true;
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debugLog("Waiting for decoder thread to exit...");
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SDL_WaitThread(decoderThread, NULL);
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decoderThread = NULL;
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}
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#endif
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}
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void OggDecoder::detachDecoder(OggDecoder *ogg)
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{
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#ifdef BBGE_BUILD_SDL
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if(decoderThread)
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{
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ogg->thread = true;
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decoderQ.push(ogg);
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}
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#endif
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}
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void OggDecoder::decode_loop(OggDecoder *this_)
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{
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while (!this_->stop_thread)
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{
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#ifdef BBGE_BUILD_SDL
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SDL_Delay(10);
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#endif
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// Transfer decoder to this background thread
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OggDecoder *ogg;
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while(decoderQ.pop(ogg))
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decoderList.push_back(ogg);
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for(std::list<OggDecoder*>::iterator it = decoderList.begin(); it != decoderList.end(); )
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{
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ogg = *it;
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if (ogg->playing)
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{
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int processed = 0;
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alGetSourcei(ogg->source, AL_BUFFERS_PROCESSED, &processed);
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for (int i = 0; i < processed; i++)
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{
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ogg->samples_done += BUFFER_LENGTH;
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ALuint buffer = 0;
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alSourceUnqueueBuffers(ogg->source, 1, &buffer);
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if (buffer)
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ogg->queue(buffer);
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}
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++it;
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}
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else
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{
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delete ogg;
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decoderList.erase(it++);
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}
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}
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core->dbg_numThreadDecoders = decoderList.size();
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}
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}
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OggDecoder::OggDecoder(VFILE *fp)
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{
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for (int i = 0; i < NUM_BUFFERS; i++)
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{
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buffers[i] = 0;
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}
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this->source = 0;
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this->fp = fp;
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this->data = NULL;
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this->data_size = 0;
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this->data_pos = 0;
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this->thread = false;
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this->playing = false;
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this->loop = false;
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this->eof = false;
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this->samples_done = 0;
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this->stopped = false;
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this->format = 0;
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this->channels = 0;
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this->forcemono = false;
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}
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OggDecoder::OggDecoder(const void *data, long data_size)
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{
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for (int i = 0; i < NUM_BUFFERS; i++)
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{
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buffers[i] = 0;
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}
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this->source = 0;
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this->fp = NULL;
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this->data = (const char *)data;
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this->data_size = data_size;
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this->data_pos = 0;
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this->thread = false;
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this->playing = false;
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this->loop = false;
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this->eof = false;
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this->samples_done = 0;
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this->stopped = false;
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this->format = 0;
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this->channels = 0;
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this->forcemono = false;
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}
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OggDecoder::~OggDecoder()
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{
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_stop();
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for (int i = 0; i < NUM_BUFFERS; i++)
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{
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if (buffers[i])
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alDeleteBuffers(1, &buffers[i]);
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}
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}
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bool OggDecoder::preStart(ALuint source)
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{
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this->source = source;
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if (fp) {
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if (ov_open_callbacks(fp, &vf, NULL, 0, local_OV_CALLBACKS_NOCLOSE) != 0)
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{
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debugLog("ov_open() failed for file");
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return false;
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}
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}
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else
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{
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data_pos = 0;
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if (ov_open_callbacks(this, &vf, NULL, 0, ogg_memory_callbacks) != 0)
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{
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debugLog("ov_open() failed for memory buffer");
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return false;
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}
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}
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vorbis_info *info = ov_info(&vf, -1);
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if (!info)
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{
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debugLog("ov_info() failed");
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ov_clear(&vf);
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return false;
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}
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channels = info->channels;
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if (info->channels == 1)
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format = AL_FORMAT_MONO16;
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else if (info->channels == 2)
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format = AL_FORMAT_STEREO16;
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else
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{
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std::ostringstream os;
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os << "Bad channel count " << info->channels;
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debugLog(os.str());
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ov_clear(&vf);
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return false;
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}
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freq = info->rate;
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/* NOTE: The failure to use alGetError() here and elsewhere is
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* intentional -- since alGetError() writes to a global buffer and
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* is thus not thread-safe, we can't use it either in the decoding
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* threads _or_ here in the main thread. In this case, we rely on
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* the specification that failing OpenAL calls do not modify return
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* parameters to detect failure; for functions that do not return
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* values, we have no choice but to hope for the best. (From a
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* multithreading point of view, the insistence on using a global
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* error buffer instead of returning success/failure or error codes
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* from functions is a remarkably poor design decision. Not that a
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* mere library user has much choice except to live with it...)
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* --achurch */
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buffers[0] = 0;
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alGenBuffers(NUM_BUFFERS, buffers);
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if (!buffers[0])
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{
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debugLog("Failed to generate OpenAL buffers");
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ov_clear(&vf);
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return false;
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}
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return true;
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}
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void OggDecoder::start(bool loop)
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{
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this->loop = loop;
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playing = true;
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eof = false;
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samples_done = 0;
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for (int i = 0; i < NUM_BUFFERS; i++)
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queue(buffers[i]);
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detachDecoder(this);
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}
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void OggDecoder::update()
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{
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if (!playing || thread)
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return;
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int processed = 0;
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alGetSourcei(source, AL_BUFFERS_PROCESSED, &processed);
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for (int i = 0; i < processed; i++)
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{
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samples_done += BUFFER_LENGTH;
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ALuint buffer;
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alSourceUnqueueBuffers(source, 1, &buffer);
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queue(buffer);
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}
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}
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void OggDecoder::stop()
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{
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if (thread)
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playing = false; // The background thread will take care of deletion then.
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else
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delete this;
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}
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void OggDecoder::_stop()
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{
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playing = false;
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if (stopped)
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return;
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ov_clear(&vf);
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alSourceStop(source);
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int queued = 0;
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alGetSourcei(source, AL_BUFFERS_QUEUED, &queued);
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for (int i = 0; i < queued; i++)
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{
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ALuint buffer;
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alSourceUnqueueBuffers(source, 1, &buffer);
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}
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for (int i = 0; i < NUM_BUFFERS; i++)
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{
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alDeleteBuffers(1, &buffers[i]);
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buffers[i] = 0;
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}
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stopped = true;
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}
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double OggDecoder::position()
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{
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ALint samples_played = 0;
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alGetSourcei(source, AL_SAMPLE_OFFSET, &samples_played);
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samples_played += samples_done;
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return (double)samples_played / (double)freq;
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}
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#if (defined(BBGE_BUILD_SDL) && (SDL_BYTEORDER == SDL_BIG_ENDIAN))
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#define BBGE_BIGENDIAN 1
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#else
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#define BBGE_BIGENDIAN 0
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#endif
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void OggDecoder::queue(ALuint buffer)
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{
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if (!playing || eof)
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return;
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const int channels = (format == AL_FORMAT_STEREO16 ? 2 : 1);
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const int buffer_size = BUFFER_LENGTH * channels * 2;
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int pcm_size = 0;
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bool just_looped = false; // Avoid infinite loops on empty files.
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while (pcm_size < buffer_size && !eof)
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{
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int bitstream_unused;
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const int nread = ov_read(
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&vf, pcm_buffer + pcm_size, buffer_size - pcm_size,
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/*bigendianp*/ BBGE_BIGENDIAN, /*word*/ 2, /*sgned*/ 1,
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&bitstream_unused
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);
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if (nread == 0 || nread == OV_EOF)
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{
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if (loop && !just_looped)
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{
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just_looped = true;
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samples_done = 0;
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ov_pcm_seek(&vf, 0);
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}
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else
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{
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eof = true;
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}
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}
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else if (nread == OV_HOLE)
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{
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debugLog("Warning: decompression error, data dropped");
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}
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else if (nread < 0)
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{
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std::ostringstream os;
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os << "Decompression error: " << nread;
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debugLog(os.str());
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eof = true;
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}
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else
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{
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pcm_size += nread;
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just_looped = false;
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}
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}
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if (pcm_size > 0)
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{
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ALuint fmt = format;
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if(channels == 2 && forcemono)
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{
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signed short *buf = (short*)&pcm_buffer[0];
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int numSamples = pcm_size / 2; // 16 bit samples
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int j = 0;
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for (int i = 0; i < numSamples ; i += 2)
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{
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// This is in theory not quite correct, but it doesn't add any artifacts or clipping.
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// FIXME: Seems that simple and stupid is the method of choice, then... -- FG
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buf[j++] = (buf[i] + buf[i+1]) / 2;
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}
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pcm_size = numSamples;
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fmt = AL_FORMAT_MONO16;
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}
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alBufferData(buffer, fmt, pcm_buffer, pcm_size, freq);
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alSourceQueueBuffers(source, 1, &buffer);
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}
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}
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size_t OggDecoder::mem_read(void *ptr, size_t size, size_t nmemb, void *datasource)
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{
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OggDecoder *this_ = (OggDecoder *)datasource;
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long to_read = size * nmemb;
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if (to_read > this_->data_size - this_->data_pos)
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to_read = this_->data_size - this_->data_pos;
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if (to_read < 0)
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to_read = 0;
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memcpy(ptr, this_->data + this_->data_pos, to_read);
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this_->data_pos += to_read;
|
|
return to_read / size;
|
|
}
|
|
|
|
int OggDecoder::mem_seek(void *datasource, ogg_int64_t offset, int whence)
|
|
{
|
|
OggDecoder *this_ = (OggDecoder *)datasource;
|
|
if (whence == SEEK_CUR)
|
|
offset += this_->data_pos;
|
|
else if (whence == SEEK_END)
|
|
offset += this_->data_size;
|
|
if (offset < 0)
|
|
offset = 0;
|
|
else if (offset > this_->data_size)
|
|
offset = this_->data_size;
|
|
this_->data_pos = offset;
|
|
return 0;
|
|
}
|
|
|
|
long OggDecoder::mem_tell(void *datasource)
|
|
{
|
|
OggDecoder *this_ = (OggDecoder *)datasource;
|
|
return this_->data_pos;
|
|
}
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
|
|
/* for porting purposes... */
|
|
#ifndef STUBBED
|
|
#ifndef _DEBUG
|
|
#define STUBBED(x)
|
|
#else
|
|
#define STUBBED(x) { \
|
|
static bool first_time = true; \
|
|
if (first_time) { \
|
|
first_time = false; \
|
|
fprintf(stderr, "STUBBED: %s (%s, %s:%d)\n", x, __FUNCTION__, __FILE__, __LINE__); \
|
|
} \
|
|
}
|
|
#endif
|
|
#endif
|
|
|
|
namespace FMOD {
|
|
|
|
#if _DEBUG
|
|
#ifdef _MSC_VER
|
|
#define bbgeDebugBreak _CrtDbgBreak
|
|
#elif defined(__GNUC__) && ((__i386__) || (__x86_64__))
|
|
#define bbgeDebugBreak() __asm__ __volatile__ ( "int $3\n\t" )
|
|
#else
|
|
#define bbgeDebugBreak() raise(SIGTRAP)
|
|
#endif
|
|
#define SANITY_CHECK_OPENAL_CALL() { \
|
|
const ALenum err = alGetError(); \
|
|
if (err != AL_NONE) { \
|
|
fprintf(stderr, "WARNING: OpenAL error %s:%d: 0x%X\n", \
|
|
__FILE__, __LINE__, (int) err); \
|
|
bbgeDebugBreak(); \
|
|
} \
|
|
}
|
|
#else
|
|
#define SANITY_CHECK_OPENAL_CALL()
|
|
#endif
|
|
|
|
// HACK: works fairly well without it. Annoying to be thrown into the debugger because all channels are full.
|
|
#undef SANITY_CHECK_OPENAL_CALL
|
|
#define SANITY_CHECK_OPENAL_CALL()
|
|
|
|
// simply nasty.
|
|
#define ALBRIDGE(cls,method,params,args) \
|
|
FMOD_RESULT cls::method params { \
|
|
if (!this) return FMOD_ERR_INTERNAL; \
|
|
return ((OpenAL##cls *) this)->method args; \
|
|
}
|
|
|
|
// FMOD::Sound implementation ...
|
|
|
|
class OpenALSound
|
|
{
|
|
public:
|
|
OpenALSound(VFILE *_fp, const bool _looping); // ctor for ogg streamed from file
|
|
OpenALSound(void *_data, long _size, const bool _looping); // ctor for ogg streamed from memory
|
|
OpenALSound(ALuint _bid, const bool _looping); // ctor for raw samples already assigned an opanAL buffer ID
|
|
VFILE *getFile() const { return fp; }
|
|
const void *getData() const { return data; }
|
|
long getSize() const { return size; }
|
|
bool isLooping() const { return looping; }
|
|
bool isRaw() const { return raw; }
|
|
FMOD_RESULT release();
|
|
FMOD_RESULT getFormat(FMOD_SOUND_TYPE *type, FMOD_SOUND_FORMAT *format, int *channels, int *bits);
|
|
void reference() { refcount++; }
|
|
ALuint getBufferName() const { return bid; }
|
|
int getNumChannels() const { return numChannels; }
|
|
void setNumChannels(int c) { numChannels = c; }
|
|
|
|
private:
|
|
VFILE * const fp;
|
|
void * const data; // Only used if fp==NULL
|
|
const long size; // Only used if fp==NULL
|
|
const bool looping;
|
|
int refcount;
|
|
const bool raw; // true if buffer holds raw PCM data
|
|
ALuint bid; // only used if raw == true
|
|
int numChannels;
|
|
};
|
|
|
|
OpenALSound::OpenALSound(VFILE *_fp, const bool _looping)
|
|
: fp(_fp)
|
|
, data(NULL)
|
|
, size(0)
|
|
, looping(_looping)
|
|
, refcount(1)
|
|
, raw(false)
|
|
, bid(0)
|
|
, numChannels(0)
|
|
{
|
|
}
|
|
|
|
OpenALSound::OpenALSound(void *_data, long _size, const bool _looping)
|
|
: fp(NULL)
|
|
, data(_data)
|
|
, size(_size)
|
|
, looping(_looping)
|
|
, refcount(1)
|
|
, raw(false)
|
|
, bid(0)
|
|
, numChannels(0)
|
|
{
|
|
}
|
|
|
|
OpenALSound::OpenALSound(ALuint _bid, const bool _looping)
|
|
: fp(NULL)
|
|
, data(NULL)
|
|
, size(0)
|
|
, looping(_looping)
|
|
, refcount(1)
|
|
, raw(true)
|
|
, bid(_bid)
|
|
, numChannels(0)
|
|
{
|
|
}
|
|
|
|
ALBRIDGE(Sound,release,(),())
|
|
FMOD_RESULT OpenALSound::release()
|
|
{
|
|
refcount--;
|
|
if (refcount <= 0)
|
|
{
|
|
if(raw)
|
|
{
|
|
alDeleteBuffers(1, &bid);
|
|
}
|
|
else
|
|
{
|
|
if (fp)
|
|
vfclose(fp);
|
|
else
|
|
free(data);
|
|
}
|
|
delete this;
|
|
}
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALBRIDGE(Sound,getFormat,(FMOD_SOUND_TYPE *type, FMOD_SOUND_FORMAT *format, int *channels, int *bits),(type, format, channels, bits))
|
|
FMOD_RESULT OpenALSound::getFormat(FMOD_SOUND_TYPE *type, FMOD_SOUND_FORMAT *format, int *channels, int *bits)
|
|
{
|
|
if(channels)
|
|
*channels = getNumChannels();
|
|
return FMOD_OK;
|
|
}
|
|
|
|
|
|
class OpenALChannelGroup;
|
|
|
|
class OpenALChannel
|
|
{
|
|
public:
|
|
OpenALChannel();
|
|
FMOD_RESULT setVolume(const float _volume);
|
|
FMOD_RESULT setPaused(const bool _paused, const bool setstate=true);
|
|
FMOD_RESULT setFrequency(const float _frequency);
|
|
FMOD_RESULT setPriority(int _priority);
|
|
FMOD_RESULT getPosition(unsigned int *position, FMOD_TIMEUNIT postype);
|
|
FMOD_RESULT getVolume(float *_volume);
|
|
FMOD_RESULT isPlaying(bool *isplaying);
|
|
FMOD_RESULT setChannelGroup(ChannelGroup *channelgroup);
|
|
FMOD_RESULT stop();
|
|
FMOD_RESULT setCallback(FMOD_CHANNEL_CALLBACK callback);
|
|
FMOD_RESULT getUserData(void **userdata);
|
|
FMOD_RESULT setUserData(void *userdata);
|
|
FMOD_RESULT set3DAttributes(const FMOD_VECTOR *pos, const FMOD_VECTOR *vel);
|
|
FMOD_RESULT set3DMinMaxDistance(float mindistance, float maxdistance);
|
|
FMOD_RESULT setMode(FMOD_MODE mode);
|
|
FMOD_RESULT getMode(FMOD_MODE *mode);
|
|
|
|
void setGroupVolume(const float _volume);
|
|
void setDistanceVolume(float _volume);
|
|
void setSourceName(const ALuint _sid) { sid = _sid; }
|
|
ALuint getSourceName() const { return sid; }
|
|
bool start(OpenALSound *sound);
|
|
void update();
|
|
void reacquire();
|
|
bool isInUse() const { return inuse; }
|
|
void setSound(OpenALSound *sound);
|
|
|
|
private:
|
|
void applyVolume();
|
|
|
|
ALuint sid; // source id.
|
|
float groupvolume;
|
|
float volume;
|
|
float distvolume;
|
|
bool paused;
|
|
int priority;
|
|
float frequency;
|
|
OpenALChannelGroup *group;
|
|
OpenALSound *sound;
|
|
OggDecoder *decoder;
|
|
bool inuse;
|
|
bool initial;
|
|
FMOD_CHANNEL_CALLBACK callback;
|
|
void *userdata;
|
|
FMOD_MODE _mode;
|
|
float mindist;
|
|
float maxdist;
|
|
bool relative;
|
|
};
|
|
|
|
|
|
class OpenALChannelGroup
|
|
{
|
|
public:
|
|
OpenALChannelGroup(const char *_name);
|
|
~OpenALChannelGroup();
|
|
FMOD_RESULT stop();
|
|
FMOD_RESULT addDSP(DSP *dsp, DSPConnection **connection);
|
|
FMOD_RESULT getPaused(bool *_paused);
|
|
FMOD_RESULT setPaused(const bool _paused);
|
|
FMOD_RESULT getVolume(float *_volume);
|
|
FMOD_RESULT setVolume(const float _volume);
|
|
bool attachChannel(OpenALChannel *channel);
|
|
void detachChannel(OpenALChannel *channel);
|
|
|
|
private:
|
|
const char *name;
|
|
bool paused;
|
|
int channel_count;
|
|
OpenALChannel **channels;
|
|
float volume;
|
|
};
|
|
|
|
|
|
// FMOD::Channel implementation...
|
|
|
|
OpenALChannel::OpenALChannel()
|
|
: sid(0)
|
|
, groupvolume(1.0f)
|
|
, volume(1.0f)
|
|
, distvolume(1.0f)
|
|
, paused(false)
|
|
, priority(0)
|
|
, frequency(1.0f)
|
|
, group(NULL)
|
|
, sound(NULL)
|
|
, decoder(NULL)
|
|
, inuse(false)
|
|
, initial(true)
|
|
, _mode(FMOD_DEFAULT)
|
|
, mindist(0.0f)
|
|
, maxdist(0.0f)
|
|
, relative(false)
|
|
{
|
|
}
|
|
|
|
void OpenALChannel::reacquire()
|
|
{
|
|
assert(!inuse);
|
|
inuse = true;
|
|
volume = 1.0f;
|
|
distvolume = 1.0f;
|
|
paused = true;
|
|
priority = 0;
|
|
frequency = 1.0f;
|
|
sound = NULL;
|
|
initial = true;
|
|
mindist = 0.0f;
|
|
maxdist = 0.0f;
|
|
relative = false;
|
|
}
|
|
|
|
void OpenALChannel::setGroupVolume(const float _volume)
|
|
{
|
|
groupvolume = _volume;
|
|
applyVolume();
|
|
}
|
|
|
|
bool OpenALChannel::start(OpenALSound *sound)
|
|
{
|
|
if (decoder)
|
|
delete decoder;
|
|
if (sound->isRaw())
|
|
{
|
|
alSourcei(sid, AL_BUFFER, sound->getBufferName());
|
|
alSourcei(sid, AL_LOOPING, sound->isLooping() ? AL_TRUE : AL_FALSE);
|
|
}
|
|
else
|
|
{
|
|
if (sound->getFile())
|
|
decoder = new OggDecoder(sound->getFile());
|
|
else
|
|
decoder = new OggDecoder(sound->getData(), sound->getSize());
|
|
if (!decoder->preStart(sid))
|
|
{
|
|
delete decoder;
|
|
decoder = NULL;
|
|
return false;
|
|
}
|
|
sound->setNumChannels(decoder->getNumChannels());
|
|
}
|
|
return true;
|
|
}
|
|
|
|
void OpenALChannel::update()
|
|
{
|
|
if (inuse)
|
|
{
|
|
if (decoder)
|
|
decoder->update();
|
|
ALint state = 0;
|
|
alGetSourceiv(sid, AL_SOURCE_STATE, &state);
|
|
SANITY_CHECK_OPENAL_CALL();
|
|
if (state == AL_STOPPED)
|
|
stop();
|
|
}
|
|
}
|
|
|
|
ALBRIDGE(Channel,setVolume,(float volume),(volume))
|
|
FMOD_RESULT OpenALChannel::setVolume(const float _volume)
|
|
{
|
|
volume = _volume;
|
|
applyVolume();
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALBRIDGE(Channel,getPosition,(unsigned int *position, FMOD_TIMEUNIT postype),(position,postype))
|
|
FMOD_RESULT OpenALChannel::getPosition(unsigned int *position, FMOD_TIMEUNIT postype)
|
|
{
|
|
assert(postype == FMOD_TIMEUNIT_MS);
|
|
if (decoder)
|
|
{
|
|
*position = (unsigned int) (decoder->position() * 1000.0);
|
|
}
|
|
else
|
|
{
|
|
ALfloat secs = 0.0f;
|
|
alGetSourcefv(sid, AL_SEC_OFFSET, &secs);
|
|
SANITY_CHECK_OPENAL_CALL();
|
|
*position = (unsigned int) (secs * 1000.0f);
|
|
}
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALBRIDGE(Channel,getVolume,(float *volume),(volume))
|
|
FMOD_RESULT OpenALChannel::getVolume(float *_volume)
|
|
{
|
|
*_volume = volume;
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALBRIDGE(Channel,isPlaying,(bool *isplaying),(isplaying))
|
|
FMOD_RESULT OpenALChannel::isPlaying(bool *isplaying)
|
|
{
|
|
// Apple's Mac OS X has a bug; alSourceRewind() doesn't make the sources
|
|
// AL_INITIAL again, so we have to track this ourselves. :/
|
|
if (initial)
|
|
{
|
|
*isplaying = true;
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALint state = 0;
|
|
alGetSourceiv(sid, AL_SOURCE_STATE, &state);
|
|
SANITY_CHECK_OPENAL_CALL();
|
|
*isplaying = ((state == AL_PLAYING) || (state == AL_INITIAL));
|
|
if (state == AL_PAUSED)
|
|
STUBBED("Should paused channels count as playing?"); // !!! FIXME
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALBRIDGE(Channel,setChannelGroup,(ChannelGroup *channelgroup),(channelgroup))
|
|
FMOD_RESULT OpenALChannel::setChannelGroup(ChannelGroup *_channelgroup)
|
|
{
|
|
OpenALChannelGroup *channelgroup = ((OpenALChannelGroup *) _channelgroup);
|
|
assert(channelgroup);
|
|
if (!channelgroup->attachChannel(this))
|
|
return FMOD_ERR_INTERNAL;
|
|
if ((group != NULL) && (group != channelgroup))
|
|
group->detachChannel(this);
|
|
group = channelgroup;
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALBRIDGE(Channel,setFrequency,(float frequency),(frequency))
|
|
FMOD_RESULT OpenALChannel::setFrequency(const float _frequency)
|
|
{
|
|
frequency = _frequency;
|
|
STUBBED("read the docs, verify this");
|
|
alSourcef(sid, AL_PITCH, _frequency);
|
|
SANITY_CHECK_OPENAL_CALL();
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALBRIDGE(Channel,setPaused,(bool paused),(paused))
|
|
FMOD_RESULT OpenALChannel::setPaused(const bool _paused, const bool setstate)
|
|
{
|
|
ALint state = 0;
|
|
alGetSourceiv(sid, AL_SOURCE_STATE, &state);
|
|
SANITY_CHECK_OPENAL_CALL();
|
|
if ((_paused) && (state == AL_PLAYING))
|
|
{
|
|
alSourcePause(sid);
|
|
initial = false;
|
|
SANITY_CHECK_OPENAL_CALL();
|
|
}
|
|
else if ((!_paused) && (initial || ((state == AL_INITIAL) || (state == AL_PAUSED))))
|
|
{
|
|
if (initial && decoder)
|
|
{
|
|
decoder->setForceMono(mindist || maxdist); // HACK: this is set for positional sounds.
|
|
decoder->start(sound->isLooping());
|
|
}
|
|
alSourcePlay(sid);
|
|
initial = false;
|
|
SANITY_CHECK_OPENAL_CALL();
|
|
}
|
|
|
|
if (setstate)
|
|
paused = _paused;
|
|
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALBRIDGE(Channel,setPriority,(int priority),(priority))
|
|
FMOD_RESULT OpenALChannel::setPriority(int _priority)
|
|
{
|
|
priority = _priority;
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALBRIDGE(Channel,stop,(),())
|
|
FMOD_RESULT OpenALChannel::stop()
|
|
{
|
|
if (decoder)
|
|
{
|
|
decoder->stop();
|
|
decoder = NULL;
|
|
}
|
|
alSourceStop(sid);
|
|
SANITY_CHECK_OPENAL_CALL();
|
|
alSourcei(sid, AL_BUFFER, 0);
|
|
SANITY_CHECK_OPENAL_CALL();
|
|
if (sound)
|
|
{
|
|
sound->release();
|
|
sound = NULL;
|
|
}
|
|
if (inuse && callback)
|
|
callback(this, FMOD_CHANNEL_CALLBACKTYPE_END, NULL, NULL); // HACK: commanddata missing (but they are not used by the callback)
|
|
paused = false;
|
|
inuse = false;
|
|
initial = false;
|
|
return FMOD_OK;
|
|
}
|
|
|
|
|
|
ALBRIDGE(Channel,setCallback,(FMOD_CHANNEL_CALLBACK callback),(callback))
|
|
FMOD_RESULT OpenALChannel::setCallback(FMOD_CHANNEL_CALLBACK callback)
|
|
{
|
|
this->callback = callback;
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALBRIDGE(Channel,getUserData,(void **userdata),(userdata))
|
|
FMOD_RESULT OpenALChannel::getUserData(void **userdata)
|
|
{
|
|
*userdata = this->userdata;
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALBRIDGE(Channel,setUserData,(void *userdata),(userdata))
|
|
FMOD_RESULT OpenALChannel::setUserData(void *userdata)
|
|
{
|
|
this->userdata = userdata;
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALBRIDGE(Channel,set3DAttributes,(const FMOD_VECTOR *pos, const FMOD_VECTOR *vel),(pos, vel))
|
|
FMOD_RESULT OpenALChannel::set3DAttributes(const FMOD_VECTOR *pos, const FMOD_VECTOR *vel)
|
|
{
|
|
if (pos)
|
|
{
|
|
alSource3f(sid, AL_POSITION, pos->x, pos->y, pos->z);
|
|
|
|
if(maxdist == mindist)
|
|
setDistanceVolume(1.0f);
|
|
else
|
|
{
|
|
// This is where custom distance attenuation starts.
|
|
float dx, dy, dz;
|
|
if(relative)
|
|
{
|
|
dx = pos->x;
|
|
dy = pos->y;
|
|
dz = pos->z;
|
|
}
|
|
else
|
|
{
|
|
dx = s_listenerPos.x - pos->x;
|
|
dy = s_listenerPos.y - pos->y;
|
|
dz = s_listenerPos.z - pos->z;
|
|
}
|
|
|
|
float d2 = dx*dx + dy*dy + dz*dz;
|
|
|
|
if(d2 < mindist*mindist)
|
|
setDistanceVolume(1.0f);
|
|
else if(d2 > maxdist*maxdist)
|
|
setDistanceVolume(0.0f);
|
|
else
|
|
{
|
|
// Replacement method for AL_INVERSE_DISTANCE_CLAMPED.
|
|
// The problem with this distance model is that the volume never goes down
|
|
// to 0, no matter how far sound sources are away from the listener.
|
|
// This could be fixed by using AL_LINEAR_DISTANCE_CLAMPED, but this model does not sound
|
|
// natural (as the gain/volume/decibels is a logarithmic measure).
|
|
// As a remedy, use a simplified quadratic 1D-bezier curve to model
|
|
// a decay similar to AL_INVERSE_DISTANCE_CLAMPED, but that actually reaches 0.
|
|
// (The formula is simplified, as the control points (1, 0, 0) cause some math to vanish.) -- FG
|
|
const float t = ((sqrtf(d2) - mindist) / (maxdist - mindist)); // [0 .. 1]
|
|
const float t1 = 1.0f - t;
|
|
const float a = t1 * t1;
|
|
const float w = 2.0f; // weight; the higher this is, the steeper is the initial falloff, and the slower the final decay before reaching 0
|
|
const float gain = a / (a + (2.0f * w * t * t1) + (t * t));
|
|
setDistanceVolume(gain);
|
|
}
|
|
}
|
|
}
|
|
|
|
if(vel)
|
|
alSource3f(sid, AL_VELOCITY, vel->x, vel->y, vel->z);
|
|
|
|
SANITY_CHECK_OPENAL_CALL();
|
|
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALBRIDGE(Channel,set3DMinMaxDistance,(float mindistance, float maxdistance),(mindistance, maxdistance))
|
|
FMOD_RESULT OpenALChannel::set3DMinMaxDistance(float mindistance, float maxdistance)
|
|
{
|
|
alSourcef(sid, AL_REFERENCE_DISTANCE, mindistance);
|
|
alSourcef(sid, AL_MAX_DISTANCE, maxdistance);
|
|
SANITY_CHECK_OPENAL_CALL();
|
|
mindist = mindistance;
|
|
maxdist = maxdistance;
|
|
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALBRIDGE(Channel,setMode,(FMOD_MODE mode),(mode))
|
|
FMOD_RESULT OpenALChannel::setMode(FMOD_MODE mode)
|
|
{
|
|
_mode = mode;
|
|
|
|
if(mode & FMOD_3D_HEADRELATIVE)
|
|
alSourcei(sid, AL_SOURCE_RELATIVE, AL_TRUE);
|
|
else // FMOD_3D_WORLDRELATIVE is the default according to FMOD docs
|
|
alSourcei(sid, AL_SOURCE_RELATIVE, AL_FALSE);
|
|
|
|
SANITY_CHECK_OPENAL_CALL();
|
|
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALBRIDGE(Channel,getMode,(FMOD_MODE *mode),(mode))
|
|
FMOD_RESULT OpenALChannel::getMode(FMOD_MODE *mode)
|
|
{
|
|
*mode = _mode;
|
|
return FMOD_OK;
|
|
}
|
|
|
|
void OpenALChannel::setDistanceVolume(float _volume)
|
|
{
|
|
distvolume = _volume;
|
|
applyVolume();
|
|
}
|
|
|
|
void OpenALChannel::applyVolume()
|
|
{
|
|
alSourcef(sid, AL_GAIN, volume * groupvolume * distvolume);
|
|
SANITY_CHECK_OPENAL_CALL();
|
|
}
|
|
|
|
|
|
|
|
// FMOD::ChannelGroup implementation...
|
|
|
|
OpenALChannelGroup::OpenALChannelGroup(const char *_name)
|
|
: name(NULL)
|
|
, paused(false)
|
|
, channel_count(0)
|
|
, channels(NULL)
|
|
, volume(1.0f)
|
|
{
|
|
if (_name)
|
|
{
|
|
char *buf = new char[strlen(_name) + 1];
|
|
strcpy(buf, _name);
|
|
name = buf;
|
|
}
|
|
}
|
|
|
|
OpenALChannelGroup::~OpenALChannelGroup()
|
|
{
|
|
delete[] name;
|
|
}
|
|
|
|
bool OpenALChannelGroup::attachChannel(OpenALChannel *channel)
|
|
{
|
|
channel->setGroupVolume(volume);
|
|
|
|
for (int i = 0; i < channel_count; i++)
|
|
{
|
|
if (channels[i] == channel)
|
|
return true;
|
|
}
|
|
|
|
void *ptr = realloc(channels, sizeof (OpenALChannel *) * (channel_count + 1));
|
|
if (ptr == NULL)
|
|
return false;
|
|
|
|
channels = (OpenALChannel **) ptr;
|
|
channels[channel_count++] = channel;
|
|
return true;
|
|
}
|
|
|
|
void OpenALChannelGroup::detachChannel(OpenALChannel *channel)
|
|
{
|
|
for (int i = 0; i < channel_count; i++)
|
|
{
|
|
if (channels[i] == channel)
|
|
{
|
|
if (i < (channel_count-1))
|
|
memmove(&channels[i], &channels[i+1], sizeof (OpenALChannel *) * ((channel_count - i) - 1));
|
|
channel_count--;
|
|
return;
|
|
}
|
|
}
|
|
|
|
assert(false && "Detached a channel that isn't part of the group!");
|
|
}
|
|
|
|
|
|
ALBRIDGE(ChannelGroup,addDSP,(DSP *dsp, DSPConnection **connection),(dsp,connection))
|
|
FMOD_RESULT OpenALChannelGroup::addDSP(DSP *dsp, DSPConnection **connection)
|
|
{
|
|
STUBBED("write me");
|
|
return FMOD_ERR_INTERNAL;
|
|
}
|
|
|
|
ALBRIDGE(ChannelGroup,getPaused,(bool *paused),(paused))
|
|
FMOD_RESULT OpenALChannelGroup::getPaused(bool *_paused)
|
|
{
|
|
*_paused = paused;
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALBRIDGE(ChannelGroup,getVolume,(float *volume),(volume))
|
|
FMOD_RESULT OpenALChannelGroup::getVolume(float *_volume)
|
|
{
|
|
*_volume = volume;
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALBRIDGE(ChannelGroup,setPaused,(bool paused),(paused))
|
|
FMOD_RESULT OpenALChannelGroup::setPaused(const bool _paused)
|
|
{
|
|
for (int i = 0; i < channel_count; i++)
|
|
channels[i]->setPaused(_paused, false);
|
|
paused = _paused;
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALBRIDGE(ChannelGroup,setVolume,(float volume),(volume))
|
|
FMOD_RESULT OpenALChannelGroup::setVolume(const float _volume)
|
|
{
|
|
volume = _volume;
|
|
for (int i = 0; i < channel_count; i++)
|
|
channels[i]->setGroupVolume(_volume);
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALBRIDGE(ChannelGroup,stop,(),())
|
|
FMOD_RESULT OpenALChannelGroup::stop()
|
|
{
|
|
for (int i = 0; i < channel_count; i++)
|
|
channels[i]->stop();
|
|
return FMOD_OK;
|
|
}
|
|
|
|
|
|
// FMOD::DSP implementation...
|
|
|
|
FMOD_RESULT DSP::getActive(bool *active)
|
|
{
|
|
STUBBED("write me");
|
|
*active = false;
|
|
return FMOD_ERR_INTERNAL;
|
|
}
|
|
|
|
FMOD_RESULT DSP::remove()
|
|
{
|
|
STUBBED("write me");
|
|
return FMOD_ERR_INTERNAL;
|
|
}
|
|
|
|
FMOD_RESULT DSP::setParameter(int index, float value)
|
|
{
|
|
STUBBED("write me");
|
|
return FMOD_ERR_INTERNAL;
|
|
}
|
|
|
|
|
|
void OpenALChannel::setSound(OpenALSound *_sound)
|
|
{
|
|
if(sound == _sound)
|
|
return;
|
|
|
|
if (sound)
|
|
sound->release();
|
|
|
|
sound = _sound;
|
|
|
|
if (sound)
|
|
sound->reference();
|
|
}
|
|
|
|
|
|
|
|
|
|
// FMOD::System implementation ...
|
|
|
|
class OpenALSystem
|
|
{
|
|
public:
|
|
OpenALSystem();
|
|
~OpenALSystem();
|
|
FMOD_RESULT init(int maxchannels, const FMOD_INITFLAGS flags, const void *extradriverdata);
|
|
FMOD_RESULT update();
|
|
FMOD_RESULT release();
|
|
FMOD_RESULT getVersion(unsigned int *version);
|
|
FMOD_RESULT setSpeakerMode(const FMOD_SPEAKERMODE speakermode);
|
|
FMOD_RESULT setFileSystem(FMOD_FILE_OPENCALLBACK useropen, FMOD_FILE_CLOSECALLBACK userclose, FMOD_FILE_READCALLBACK userread, FMOD_FILE_SEEKCALLBACK userseek, const int blockalign);
|
|
FMOD_RESULT setDSPBufferSize(const unsigned int bufferlength, const int numbuffers);
|
|
FMOD_RESULT createChannelGroup(const char *name, ChannelGroup **channelgroup);
|
|
FMOD_RESULT createDSPByType(const FMOD_DSP_TYPE type, DSP **dsp);
|
|
FMOD_RESULT createSound(const char *name_or_data, const FMOD_MODE mode, const FMOD_CREATESOUNDEXINFO *exinfo, Sound **sound);
|
|
FMOD_RESULT createStream(const char *name_or_data, const FMOD_MODE mode, const FMOD_CREATESOUNDEXINFO *exinfo, Sound **sound);
|
|
FMOD_RESULT getDriverCaps(const int id, FMOD_CAPS *caps, int *minfrequency, int *maxfrequency, FMOD_SPEAKERMODE *controlpanelspeakermode);
|
|
FMOD_RESULT getMasterChannelGroup(ChannelGroup **channelgroup);
|
|
FMOD_RESULT playSound(FMOD_CHANNELINDEX channelid, Sound *sound, bool paused, Channel **channel);
|
|
FMOD_RESULT set3DListenerAttributes(int listener, const FMOD_VECTOR *pos, const FMOD_VECTOR *vel, const FMOD_VECTOR *forward, const FMOD_VECTOR *up);
|
|
|
|
FMOD_RESULT getNumChannels(int *maxchannels_ret);
|
|
|
|
private:
|
|
OpenALChannelGroup *master_channel_group;
|
|
int num_channels;
|
|
OpenALChannel *channels;
|
|
};
|
|
|
|
|
|
OpenALSystem::OpenALSystem()
|
|
: master_channel_group(NULL)
|
|
, num_channels(0)
|
|
, channels(NULL)
|
|
{
|
|
}
|
|
|
|
OpenALSystem::~OpenALSystem()
|
|
{
|
|
delete master_channel_group;
|
|
delete[] channels;
|
|
}
|
|
|
|
|
|
FMOD_RESULT System_Create(FMOD_SYSTEM **system)
|
|
{
|
|
*system = (FMOD_SYSTEM *) new OpenALSystem;
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALBRIDGE(System,createChannelGroup,(const char *name, ChannelGroup **channelgroup),(name,channelgroup))
|
|
FMOD_RESULT OpenALSystem::createChannelGroup(const char *name, ChannelGroup **channelgroup)
|
|
{
|
|
*channelgroup = (ChannelGroup *) new OpenALChannelGroup(name);
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALBRIDGE(System,createDSPByType,(FMOD_DSP_TYPE type, DSP **dsp),(type,dsp))
|
|
FMOD_RESULT OpenALSystem::createDSPByType(const FMOD_DSP_TYPE type, DSP **dsp)
|
|
{
|
|
*dsp = NULL;
|
|
STUBBED("write me");
|
|
return FMOD_ERR_INTERNAL;
|
|
}
|
|
|
|
static void *decode_to_pcm(VFILE *io, ALenum &format, ALsizei &size, ALuint &freq)
|
|
{
|
|
ALubyte *retval = NULL;
|
|
|
|
// Uncompress and feed to the AL.
|
|
OggVorbis_File vf;
|
|
memset(&vf, '\0', sizeof (vf));
|
|
if (ov_open_callbacks(io, &vf, NULL, 0, local_OV_CALLBACKS_NOCLOSE) == 0)
|
|
{
|
|
int bitstream = 0;
|
|
vorbis_info *info = ov_info(&vf, -1);
|
|
size = 0;
|
|
format = (info->channels == 1) ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16;
|
|
freq = info->rate;
|
|
|
|
if ((info->channels != 1) && (info->channels != 2))
|
|
{
|
|
ov_clear(&vf);
|
|
return NULL;
|
|
}
|
|
|
|
char buf[1024 * 16];
|
|
long rc = 0;
|
|
size_t allocated = 64 * 1024;
|
|
retval = (ALubyte *) malloc(allocated);
|
|
while ( (rc = ov_read(&vf, buf, sizeof (buf), BBGE_BIGENDIAN, 2, 1, &bitstream)) != 0 )
|
|
{
|
|
if (rc > 0)
|
|
{
|
|
size += rc;
|
|
if (size >= allocated)
|
|
{
|
|
allocated *= 2;
|
|
ALubyte *tmp = (ALubyte *) realloc(retval, allocated);
|
|
if (tmp == NULL)
|
|
{
|
|
free(retval);
|
|
retval = NULL;
|
|
break;
|
|
}
|
|
retval = tmp;
|
|
}
|
|
memcpy(retval + (size - rc), buf, rc);
|
|
}
|
|
}
|
|
ov_clear(&vf);
|
|
return retval;
|
|
}
|
|
return NULL;
|
|
}
|
|
|
|
|
|
ALBRIDGE(System,createSound,(const char *name_or_data, FMOD_MODE mode, FMOD_CREATESOUNDEXINFO *exinfo, Sound **sound),(name_or_data,mode,exinfo,sound))
|
|
FMOD_RESULT OpenALSystem::createSound(const char *name_or_data, const FMOD_MODE mode, const FMOD_CREATESOUNDEXINFO *exinfo, Sound **sound)
|
|
{
|
|
assert(!exinfo);
|
|
|
|
FMOD_RESULT retval = FMOD_ERR_INTERNAL;
|
|
|
|
// !!! FIXME: if it's not Ogg, we don't have a decoder. I'm lazy. :/
|
|
char *fname = (char *) alloca(strlen(name_or_data) + 16);
|
|
strcpy(fname, name_or_data);
|
|
char *ptr = strrchr(fname, '.');
|
|
if (ptr) *ptr = '\0';
|
|
strcat(fname, ".ogg");
|
|
|
|
// just in case...
|
|
VFILE *io = vfopen(core->adjustFilenameCase(fname).c_str(), "rb");
|
|
if (io == NULL)
|
|
return FMOD_ERR_INTERNAL;
|
|
|
|
if (mode & FMOD_CREATESTREAM)
|
|
{
|
|
// Create streaming file handle decoder
|
|
*sound = (Sound *) new OpenALSound(io, (((mode & FMOD_LOOP_OFF) == 0) && (mode & FMOD_LOOP_NORMAL)));
|
|
retval = FMOD_OK;
|
|
}
|
|
else if(core->settings.prebufferSounds)
|
|
{
|
|
// Pre-decode the sound file and store the raw PCM buffer
|
|
ALenum format = AL_NONE;
|
|
ALsizei size = 0;
|
|
ALuint freq = 0;
|
|
void *data = decode_to_pcm(io, format, size, freq);
|
|
vfclose(io);
|
|
|
|
ALuint bid = 0;
|
|
alGenBuffers(1, &bid);
|
|
if (bid != 0)
|
|
{
|
|
// FIXME: This needs to stored seperately and fed to the AL on demand,
|
|
// converting stereo to mono when it's known whether to do so or not.
|
|
alBufferData(bid, format, data, size, freq);
|
|
*sound = (Sound *) new OpenALSound(bid, (((mode & FMOD_LOOP_OFF) == 0) && (mode & FMOD_LOOP_NORMAL)));
|
|
((OpenALSound*)*sound)->setNumChannels(format == AL_FORMAT_STEREO16 ? 2 : 1);
|
|
retval = FMOD_OK;
|
|
}
|
|
free(data);
|
|
}
|
|
else
|
|
{
|
|
// Create streaming memory decoder
|
|
vfseek(io, 0, SEEK_END);
|
|
long size = vftell(io);
|
|
if (vfseek(io, 0, SEEK_SET) != 0)
|
|
{
|
|
debugLog("Seek error on " + std::string(fname));
|
|
vfclose(io);
|
|
return FMOD_ERR_INTERNAL;
|
|
}
|
|
|
|
void *data = malloc(size);
|
|
if (data == NULL)
|
|
{
|
|
debugLog("Out of memory for " + std::string(fname));
|
|
vfclose(io);
|
|
return FMOD_ERR_INTERNAL;
|
|
}
|
|
|
|
long nread = vfread(data, 1, size, io);
|
|
vfclose(io);
|
|
if (nread != size)
|
|
{
|
|
debugLog("Failed to read data from " + std::string(fname));
|
|
free(data);
|
|
return FMOD_ERR_INTERNAL;
|
|
}
|
|
|
|
*sound = (Sound *) new OpenALSound(data, size, (((mode & FMOD_LOOP_OFF) == 0) && (mode & FMOD_LOOP_NORMAL)));
|
|
retval = FMOD_OK;
|
|
}
|
|
|
|
return retval;
|
|
}
|
|
|
|
ALBRIDGE(System,createStream,(const char *name_or_data, FMOD_MODE mode, FMOD_CREATESOUNDEXINFO *exinfo, Sound **sound),(name_or_data,mode,exinfo,sound))
|
|
FMOD_RESULT OpenALSystem::createStream(const char *name_or_data, const FMOD_MODE mode, const FMOD_CREATESOUNDEXINFO *exinfo, Sound **sound)
|
|
{
|
|
return createSound(name_or_data, mode | FMOD_CREATESTREAM, exinfo, sound);
|
|
}
|
|
|
|
ALBRIDGE(System,getDriverCaps,(int id, FMOD_CAPS *caps, int *minfrequency, int *maxfrequency, FMOD_SPEAKERMODE *controlpanelspeakermode),(id,caps,minfrequency,maxfrequency,controlpanelspeakermode))
|
|
FMOD_RESULT OpenALSystem::getDriverCaps(const int id, FMOD_CAPS *caps, int *minfrequency, int *maxfrequency, FMOD_SPEAKERMODE *controlpanelspeakermode)
|
|
{
|
|
assert(!id);
|
|
assert(!minfrequency);
|
|
assert(!maxfrequency);
|
|
*controlpanelspeakermode = FMOD_SPEAKERMODE_STEREO; // not strictly true, but works for aquaria's usage.
|
|
*caps = 0; // aquaria only checks FMOD_CAPS_HARDWARE_EMULATED.
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALBRIDGE(System,getMasterChannelGroup,(ChannelGroup **channelgroup),(channelgroup))
|
|
FMOD_RESULT OpenALSystem::getMasterChannelGroup(ChannelGroup **channelgroup)
|
|
{
|
|
*channelgroup = (ChannelGroup *) master_channel_group;
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALBRIDGE(System,getVersion,(unsigned int *version),(version))
|
|
FMOD_RESULT OpenALSystem::getVersion(unsigned int *version)
|
|
{
|
|
*version = FMOD_VERSION;
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALBRIDGE(System,init,(int maxchannels, FMOD_INITFLAGS flags, void *extradriverdata),(maxchannels,flags,extradriverdata))
|
|
FMOD_RESULT OpenALSystem::init(int maxchannels, const FMOD_INITFLAGS flags, const void *extradriverdata)
|
|
{
|
|
ALCdevice *dev = alcOpenDevice(NULL);
|
|
if (!dev)
|
|
return FMOD_ERR_INTERNAL;
|
|
|
|
// OpenAL doesn't provide a way to request sources that can be either
|
|
// mono or stereo, so we need to request both separately (thus allocating
|
|
// twice the theoretical requirement -- oh well). --achurch
|
|
ALCint requested_attributes[5];
|
|
requested_attributes[0] = ALC_MONO_SOURCES;
|
|
requested_attributes[1] = maxchannels;
|
|
requested_attributes[2] = ALC_STEREO_SOURCES;
|
|
requested_attributes[3] = maxchannels;
|
|
requested_attributes[4] = 0;
|
|
ALCcontext *ctx = alcCreateContext(dev, requested_attributes);
|
|
if (!ctx)
|
|
{
|
|
alcCloseDevice(dev);
|
|
return FMOD_ERR_INTERNAL;
|
|
}
|
|
|
|
ALCint num_attributes = 0;
|
|
alcGetIntegerv(dev, ALC_ATTRIBUTES_SIZE, 1, &num_attributes);
|
|
if (num_attributes > 0) {
|
|
ALCint *attributes = new ALCint[num_attributes];
|
|
alcGetIntegerv(dev, ALC_ALL_ATTRIBUTES, num_attributes, attributes);
|
|
int i;
|
|
for (i = 0; i < num_attributes; i += 2) {
|
|
if (attributes[i] == ALC_MONO_SOURCES) {
|
|
if (attributes[i+1] <= 0) {
|
|
debugLog("Couldn't get any mono sources, aborting");
|
|
alcDestroyContext(ctx);
|
|
alcCloseDevice(dev);
|
|
return FMOD_ERR_INTERNAL;
|
|
} else if (attributes[i+1] < num_channels) {
|
|
std::ostringstream os;
|
|
os << "Only got " << attributes[i+1] << " of "
|
|
<< maxchannels << " mono sources";
|
|
debugLog(os.str());
|
|
maxchannels = attributes[i+1];
|
|
}
|
|
} else if (attributes[i] == ALC_STEREO_SOURCES) {
|
|
if (attributes[i+1] <= 0) {
|
|
debugLog("Couldn't get any stereo sources, aborting");
|
|
alcDestroyContext(ctx);
|
|
alcCloseDevice(dev);
|
|
return FMOD_ERR_INTERNAL;
|
|
} else if (attributes[i+1] < num_channels) {
|
|
std::ostringstream os;
|
|
os << "Only got " << attributes[i+1] << " of "
|
|
<< maxchannels << " stereo sources";
|
|
debugLog(os.str());
|
|
maxchannels = attributes[i+1];
|
|
}
|
|
}
|
|
}
|
|
delete[] attributes;
|
|
} else {
|
|
debugLog("WARNING: couldn't get device attributes!");
|
|
}
|
|
|
|
alcMakeContextCurrent(ctx);
|
|
alcProcessContext(ctx);
|
|
|
|
#ifdef _DEBUG
|
|
printf("AL_VENDOR: %s\n", (char *) alGetString(AL_VENDOR));
|
|
printf("AL_RENDERER: %s\n", (char *) alGetString(AL_RENDERER));
|
|
printf("AL_VERSION: %s\n", (char *) alGetString(AL_VERSION));
|
|
printf("AL_EXTENSIONS: %s\n", (char *) alGetString(AL_EXTENSIONS));
|
|
#endif
|
|
|
|
SANITY_CHECK_OPENAL_CALL();
|
|
|
|
master_channel_group = new OpenALChannelGroup("master");
|
|
|
|
num_channels = maxchannels;
|
|
channels = new OpenALChannel[maxchannels];
|
|
ALenum err = alGetError(); // clear any existing error state.
|
|
for (int i = 0; i < num_channels; i++)
|
|
{
|
|
ALuint sid = 0;
|
|
alGenSources(1, &sid);
|
|
err = alGetError();
|
|
if (err != AL_NONE)
|
|
{
|
|
char errmsg[512];
|
|
sprintf(errmsg, "WARNING: OpenAL error %s:%d: 0x%X\n", __FILE__, __LINE__, (int) err);
|
|
debugLog(errmsg);
|
|
num_channels = i - 1; // last channel that worked
|
|
break;
|
|
}
|
|
|
|
alSourcei(sid, AL_SOURCE_RELATIVE, AL_FALSE);
|
|
SANITY_CHECK_OPENAL_CALL();
|
|
alSource3f(sid, AL_POSITION, 0.0f, 0.0f, 0.0f); // no panning or spatialization in Aquaria.
|
|
SANITY_CHECK_OPENAL_CALL();
|
|
channels[i].setSourceName(sid);
|
|
channels[i].setChannelGroup((ChannelGroup *) master_channel_group);
|
|
}
|
|
std::stringstream ss;
|
|
ss << "Using " << num_channels << " sound channels.";
|
|
debugLog(ss.str());
|
|
|
|
// HACK: FMOD doesn't do this.
|
|
// For completeness, we pass FMOD_3D_LINEARROLLOFF to createSound().
|
|
// We do our own non-standard distance attenuation model, as the modes offered by OpenAL
|
|
// are not sufficient. See OpenALChannel::set3DMinMaxDistance() for the gain control code. -- FG
|
|
alDistanceModel(AL_NONE);
|
|
SANITY_CHECK_OPENAL_CALL();
|
|
s_listenerPos.x = s_listenerPos.y = s_listenerPos.z = 0.0f;
|
|
|
|
|
|
OggDecoder::startDecoderThread();
|
|
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALBRIDGE(System,getNumChannels,(int *maxchannels_ret),(maxchannels_ret))
|
|
FMOD_RESULT OpenALSystem::getNumChannels(int *maxchannels_ret)
|
|
{
|
|
*maxchannels_ret = num_channels;
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALBRIDGE(System,playSound,(FMOD_CHANNELINDEX channelid, Sound *sound, bool paused, Channel **channel),(channelid,sound,paused,channel))
|
|
FMOD_RESULT OpenALSystem::playSound(FMOD_CHANNELINDEX channelid, Sound *_sound, bool paused, Channel **channel)
|
|
{
|
|
*channel = NULL;
|
|
|
|
if (channelid == FMOD_CHANNEL_FREE)
|
|
{
|
|
for (int i = 0; i < num_channels; i++)
|
|
{
|
|
if (!channels[i].isInUse())
|
|
{
|
|
channelid = (FMOD_CHANNELINDEX) i;
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
if ((channelid < 0) || (channelid >= num_channels))
|
|
return FMOD_ERR_INTERNAL;
|
|
|
|
OpenALSound *sound = (OpenALSound *) _sound;
|
|
const ALuint sid = channels[channelid].getSourceName();
|
|
// alSourceRewind doesn't work right on some versions of Mac OS X.
|
|
alSourceStop(sid); // stop any playback, set to AL_INITIAL.
|
|
alSourceRewind(sid); // stop any playback, set to AL_INITIAL.
|
|
SANITY_CHECK_OPENAL_CALL();
|
|
alSourcei(sid, AL_BUFFER, 0); // Reset state to AL_UNDETERMINED.
|
|
SANITY_CHECK_OPENAL_CALL();
|
|
|
|
if (!channels[channelid].start(sound))
|
|
return FMOD_ERR_INTERNAL;
|
|
|
|
channels[channelid].reacquire();
|
|
channels[channelid].setPaused(paused);
|
|
channels[channelid].setSound(sound);
|
|
*channel = (Channel *) &channels[channelid];
|
|
return FMOD_OK;
|
|
}
|
|
|
|
|
|
ALBRIDGE(System,release,(),())
|
|
FMOD_RESULT OpenALSystem::release()
|
|
{
|
|
OggDecoder::stopDecoderThread();
|
|
|
|
ALCcontext *ctx = alcGetCurrentContext();
|
|
if (ctx)
|
|
{
|
|
for (int i = 0; i < num_channels; i++)
|
|
{
|
|
const ALuint sid = channels[i].getSourceName();
|
|
channels[i].setSourceName(0);
|
|
channels[i].setSound(NULL);
|
|
alSourceStop(sid);
|
|
alSourcei(sid, AL_BUFFER, 0);
|
|
alDeleteSources(1, &sid);
|
|
}
|
|
ALCdevice *dev = alcGetContextsDevice(ctx);
|
|
alcMakeContextCurrent(NULL);
|
|
alcSuspendContext(ctx);
|
|
alcDestroyContext(ctx);
|
|
alcCloseDevice(dev);
|
|
}
|
|
delete this;
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALBRIDGE(System,setDSPBufferSize,(unsigned int bufferlength, int numbuffers),(bufferlength,numbuffers))
|
|
FMOD_RESULT OpenALSystem::setDSPBufferSize(const unsigned int bufferlength, const int numbuffers)
|
|
{
|
|
// aquaria only uses this for FMOD_CAPS_HARDWARE_EMULATED, so I skipped it.
|
|
return FMOD_ERR_INTERNAL;
|
|
}
|
|
|
|
ALBRIDGE(System,setFileSystem,(FMOD_FILE_OPENCALLBACK useropen, FMOD_FILE_CLOSECALLBACK userclose, FMOD_FILE_READCALLBACK userread, FMOD_FILE_SEEKCALLBACK userseek, int blockalign),(useropen,userclose,userread,userseek,blockalign))
|
|
FMOD_RESULT OpenALSystem::setFileSystem(FMOD_FILE_OPENCALLBACK useropen, FMOD_FILE_CLOSECALLBACK userclose, FMOD_FILE_READCALLBACK userread, FMOD_FILE_SEEKCALLBACK userseek, const int blockalign)
|
|
{
|
|
// Aquaria sets these, but they don't do anything fancy, so we ignore them for now.
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALBRIDGE(System,setSpeakerMode,(FMOD_SPEAKERMODE speakermode),(speakermode))
|
|
FMOD_RESULT OpenALSystem::setSpeakerMode(const FMOD_SPEAKERMODE speakermode)
|
|
{
|
|
return FMOD_OK; // we ignore this for Aquaria.
|
|
}
|
|
|
|
ALBRIDGE(System,update,(),())
|
|
FMOD_RESULT OpenALSystem::update()
|
|
{
|
|
alcProcessContext(alcGetCurrentContext());
|
|
for (int i = 0; i < num_channels; i++)
|
|
channels[i].update();
|
|
#if _DEBUG
|
|
const ALenum err = alGetError();
|
|
if (err != AL_NONE)
|
|
fprintf(stderr, "WARNING: OpenAL error this frame: 0x%X\n", (int) err);
|
|
#endif
|
|
return FMOD_OK;
|
|
}
|
|
|
|
ALBRIDGE(System, set3DListenerAttributes, (int listener, const FMOD_VECTOR *pos, const FMOD_VECTOR *vel, const FMOD_VECTOR *forward, const FMOD_VECTOR *up),
|
|
(listener, pos, vel, forward, up));
|
|
FMOD_RESULT OpenALSystem::set3DListenerAttributes(int listener, const FMOD_VECTOR *pos, const FMOD_VECTOR *vel, const FMOD_VECTOR *forward, const FMOD_VECTOR *up)
|
|
{
|
|
// ignore listener parameter; there is only one listener in OpenAL.
|
|
|
|
if(up || forward)
|
|
{
|
|
ALfloat orientation[6];
|
|
alGetListenerfv(AL_ORIENTATION, &orientation[0]);
|
|
|
|
if(forward)
|
|
{
|
|
orientation[0] = forward->x;
|
|
orientation[1] = forward->y;
|
|
orientation[2] = forward->z;
|
|
}
|
|
|
|
if(up)
|
|
{
|
|
orientation[3] = up->x;
|
|
orientation[4] = up->y;
|
|
orientation[5] = up->z;
|
|
}
|
|
|
|
alListenerfv(AL_ORIENTATION, &orientation[0]);
|
|
}
|
|
|
|
if(pos)
|
|
{
|
|
s_listenerPos = *pos;
|
|
alListener3f(AL_POSITION, pos->x, pos->y, pos->z);
|
|
}
|
|
|
|
if(vel)
|
|
alListener3f(AL_VELOCITY, vel->x, vel->y, vel->z);
|
|
|
|
return FMOD_OK;
|
|
}
|
|
|
|
|
|
// misc FMOD bits...
|
|
|
|
FMOD_RESULT Memory_GetStats(int *currentalloced, int *maxalloced, FMOD_BOOL blocking)
|
|
{
|
|
// not ever used by Aquaria.
|
|
*currentalloced = *maxalloced = 42;
|
|
return FMOD_ERR_INTERNAL;
|
|
}
|
|
|
|
} // namespace FMOD
|
|
|
|
// end of FmodOpenALBridge.cpp ...
|
|
|