warzone2100/lib/sound/openal_track.c

/*
	This file is part of Warzone 2100.
	Copyright (C) 1999-2004  Eidos Interactive
	Copyright (C) 2005-2007  Warzone Resurrection Project

	Warzone 2100 is free software; you can redistribute it and/or modify
	it under the terms of the GNU General Public License as published by
	the Free Software Foundation; either version 2 of the License, or
	(at your option) any later version.

	Warzone 2100 is distributed in the hope that it will be useful,
	but WITHOUT ANY WARRANTY; without even the implied warranty of
	MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
	GNU General Public License for more details.

	You should have received a copy of the GNU General Public License
	along with Warzone 2100; if not, write to the Free Software
	Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
*/
//*
//
// Sound library-specific functions
//*
//

// this has to be first
#include "lib/framework/frame.h"
#include "lib/framework/frameresource.h"

#ifndef WZ_NOSOUND
# ifdef WZ_OS_MAC
#  include <OpenAL/al.h>
#  include <OpenAL/alc.h>
# else
#  include <AL/al.h>
#  include <AL/alc.h>
# endif
#endif

#include <physfs.h>
#include <string.h>

#include "tracklib.h"
#include "audio.h"
#include "cdaudio.h"
#include "oggvorbis.h"
#include "openal_error.h"

#define ATTENUATION_FACTOR	0.0003f

#ifndef WZ_NOSOUND
ALuint current_queue_sample = -1;
#endif

typedef struct	SAMPLE_LIST
{
	struct AUDIO_SAMPLE *curr;
	struct SAMPLE_LIST	*next;
} SAMPLE_LIST;

static SAMPLE_LIST *active_samples = NULL;

#ifndef WZ_NOSOUND
static ALfloat		sfx_volume = 1.0;
static ALfloat		sfx3d_volume = 1.0;

static ALCdevice* device = 0;
static ALCcontext* context = 0;
#endif

BOOL openal_initialized = FALSE;

/** Removes the given sample from the "active_samples" linked list
 *  \param previous either NULL (if \c to_remove is the first item in the
 *                  list) or the item occurring just before \c to_remove in
 *                  the list
 *  \param to_remove the item to actually remove from the list
 */
static void sound_RemoveSample(SAMPLE_LIST* previous, SAMPLE_LIST* to_remove)
{
	if (previous != NULL && previous != to_remove)
	{
		// Verify that the given two samples actually follow eachother in the list
		ASSERT(previous->next == to_remove, "Sound samples don't follow eachother in the list, we're probably removing the wrong item.");

		// Remove the item to remove from the linked list by skipping
		// it in the pointer sequence.
		previous->next = to_remove->next;
	}
	else
	{
		// Apparently we're removing the first item from the list. So
		// make the next one the list's head.
		active_samples = to_remove->next;
	}
}

#ifndef WZ_NOSOUND
static void PrintOpenALVersion(code_part part)
{
	debug(part, "OpenAL Vendor: %s", alGetString(AL_VENDOR));
	debug(part, "OpenAL Version: %s", alGetString(AL_VERSION));
	debug(part, "OpenAL Renderer: %s", alGetString(AL_RENDERER));
	debug(part, "OpenAL Extensions: %s", alGetString(AL_EXTENSIONS));
}
#endif

//*
// =======================================================================================================================
// =======================================================================================================================
//
BOOL sound_InitLibrary( void )
{
#ifndef WZ_NOSOUND
	int err;
	ALfloat listenerVel[3] = { 0.0, 0.0, 0.0 };
	ALfloat listenerOri[6] = { 0.0, 0.0, 1.0, 0.0, 1.0, 0.0 };

	device = alcOpenDevice(0);
	if(device == 0)
	{
		PrintOpenALVersion(LOG_ERROR);
		debug(LOG_ERROR, "Couldn't open audio device.");
		return FALSE;
	}

	context = alcCreateContext(device, NULL);		//NULL was contextAttributes
	alcMakeContextCurrent(context);

	err = sound_GetDeviceError(device);
	if (err != ALC_NO_ERROR)
	{
		PrintOpenALVersion(LOG_ERROR);
		debug(LOG_ERROR, "Couldn't initialize audio context: %s", alcGetString(device, err));
		return FALSE;
	}
#endif

	openal_initialized = TRUE;

#ifndef WZ_NOSOUND
	// Clear Error Codes
	alGetError();
	alcGetError(device);

	// Check what version of Open AL we are using
	PrintOpenALVersion(LOG_SOUND);


	alListener3f(AL_POSITION, 0.f, 0.f, 0.f);
	alListenerfv( AL_VELOCITY, listenerVel );
	alListenerfv( AL_ORIENTATION, listenerOri );
	alDistanceModel( AL_NONE );
#endif
	return TRUE;
}

//*
// =======================================================================================================================
// =======================================================================================================================
//
void sound_ShutdownLibrary( void )
{
	SAMPLE_LIST * aSample = active_samples, * tmpSample = NULL;

	debug(LOG_SOUND, "sound_ShutdownLibrary: starting shutdown");
#ifndef WZ_NOSOUND
	if(context != 0) {
#ifdef WIN32
		/* Ifdef'ed out the two lines below on Linux since this caused some versions
		 * of OpenAL to hang on exit. - Per */
		debug(LOG_SOUND, "sound_ShutdownLibrary: make default context NULL");
		alcMakeContextCurrent(NULL);		//this should work now -Q
#endif
		debug(LOG_SOUND, "sound_ShutdownLibrary: destroy previous context");
		alcDestroyContext(context); // this gives a long delay on some impl.
		context = 0;
	}
	debug(LOG_SOUND, "sound_ShutdownLibrary: close device");
	if(device != 0) {
		alcCloseDevice(device);
		device = 0;
	}
#endif

	while( aSample )
	{
		tmpSample = aSample->next;
		free( aSample );
		aSample = tmpSample;
	}
	active_samples = NULL;
}

//*
// =======================================================================================================================
// =======================================================================================================================
//
void sound_Update( void )
{
#ifndef WZ_NOSOUND
	int err;
#endif
	SAMPLE_LIST* node = active_samples;
	SAMPLE_LIST* previous = NULL;

	while (node != NULL)
	{
#ifndef WZ_NOSOUND
		ALenum state;

		alGetSourcei(node->curr->iSample, AL_SOURCE_STATE, &state);

		// Check whether an error occurred while retrieving the state.
		// If one did, the state returned is useless. So instead of
		// using it continue with the next sample.
		err = sound_GetError();
		if (err != AL_NO_ERROR)
			continue;

		switch (state)
		{
			case AL_PLAYING:
			case AL_PAUSED:
				// If we haven't finished playing yet, just
				// continue with the next item in the list.

				// sound_SetObjectPosition(i->curr->iSample, i->curr->x, i->curr->y, i->curr->z);

				// Move to the next object
				previous = node;
				node = node->next;
				break;

			case AL_STOPPED:
#endif
				sound_FinishedCallback(node->curr);
#ifndef WZ_NOSOUND

			default:
				// If an OpenAL source is associated with this sample, release it
				if (node->curr->iSample != (ALuint)AL_INVALID)
				{
					alDeleteSources(1, &node->curr->iSample);
					sound_GetError();
				}
#endif
				// Remove the sample from the list
				sound_RemoveSample(previous, node);
				// Free it
				free(node);
				
				// Get a pointer to the next node, the previous pointer doesn't change
				node = previous->next;
#ifndef WZ_NOSOUND
				break;
		}
#endif
	}

	cdAudio_Update();

#ifndef WZ_NOSOUND
	// Reset the current error state
	alcGetError(device);

	alcProcessContext(context);

	err = sound_GetDeviceError(device);
	if (err != ALC_NO_ERROR)
	{
		debug(LOG_ERROR, "Error while processing audio context: %s",
		      alGetString(err));
	}
#endif
	}

//*
// =======================================================================================================================
// =======================================================================================================================
//
BOOL sound_QueueSamplePlaying( void )
{
#ifndef WZ_NOSOUND
	if ( current_queue_sample == (ALuint)AL_INVALID )
	{
		return FALSE;
	}
		
	ALenum	state;

	alGetSourcei(current_queue_sample, AL_SOURCE_STATE, &state);

	// Check whether an error occurred while retrieving the state.
	// If one did, the state returned is useless. So instead of
	// using it return false.
	if (sound_GetError() != AL_NO_ERROR)
		return FALSE;

	if (state == AL_PLAYING)
	{
		return TRUE;
	}

	if (current_queue_sample != (ALuint)AL_INVALID)
	{
		alDeleteSources(1, &current_queue_sample);
		sound_GetError();
		current_queue_sample = AL_INVALID;
	}
#endif
	return FALSE;
}

/** Decodes an opened OggVorbis file into an OpenAL buffer
 *  \param psTrack pointer to object which will contain the final buffer
 *  \param PHYSFS_fileHandle file handle given by PhysicsFS to the opened file
 *  \return on success the psTrack pointer, otherwise it will be free'd and a NULL pointer is returned instead
 */
static inline TRACK* sound_DecodeOggVorbisTrack(TRACK *psTrack, PHYSFS_file* PHYSFS_fileHandle)
{
#ifndef WZ_NOSOUND
	ALenum		format;
	ALuint		buffer;

	struct OggVorbisDecoderState* decoder = sound_CreateOggVorbisDecoder(PHYSFS_fileHandle, TRUE);
	soundDataBuffer* soundBuffer;

	soundBuffer = sound_DecodeOggVorbis(decoder, 0);
	sound_DestroyOggVorbisDecoder(decoder);

	if (soundBuffer == NULL)
	{
		free(psTrack);
		return NULL;
	}

	// Determine PCM data format
	format = (soundBuffer->channelCount == 1) ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16;

	// Create an OpenAL buffer and fill it with the decoded data
	alGenBuffers(1, &buffer);
	sound_GetError();
	alBufferData(buffer, format, soundBuffer->data, soundBuffer->size, soundBuffer->frequency);
	sound_GetError();

	free(soundBuffer);

	// save buffer name in track
	psTrack->iBufferName = buffer;
#endif

	return psTrack;
}

//*
// =======================================================================================================================
// =======================================================================================================================
//
TRACK* sound_LoadTrackFromFile(const char *fileName)
{
	TRACK* pTrack;
	PHYSFS_file* fileHandle;
	size_t filename_size;

	// Use PhysicsFS to open the file
	fileHandle = PHYSFS_openRead(fileName);
	if (fileHandle == NULL)
	{
		debug(LOG_ERROR, "sound_LoadTrackFromFile: PHYSFS_openRead(\"%s\") failed with error: %s\n", fileName, PHYSFS_getLastError());
		return NULL;
	}

	if (GetLastResourceFilename() == NULL)
	{
		// This is a non fatal error.  We just can't find filename for some reason.
		debug(LOG_WARNING, "sound_LoadTrackFromFile: missing resource filename?");
		filename_size = 0;
	}
	else
	{
		filename_size = strlen(GetLastResourceFilename()) + 1;
	}

	// allocate track, plus the memory required to contain the filename
	// one malloc call ensures only one free call is required
	pTrack = (TRACK*)malloc(sizeof(TRACK) + filename_size);
	if (pTrack == NULL)
	{
		debug( LOG_ERROR, "sound_ConstructTrack: couldn't allocate memory\n" );
		abort();
		return NULL;
	}

	// Initialize everyting (except for the filename) to zero
	memset(pTrack, 0, sizeof(TRACK));

	// Set filename pointer; if the filename (as returned by
	// GetLastResourceFilename()) is a NULL pointer, then this will be a
	// NULL pointer as well.
	pTrack->fileName = filename_size ? (const char*)pTrack + sizeof(TRACK) : NULL;

	// Copy the filename into the struct, if we don't have a NULL pointer
	if (filename_size != 0)
	{
		strcpy((char*)pTrack->fileName, GetLastResourceFilename());
	}

	// Now use sound_ReadTrackFromBuffer to decode the file's contents
	pTrack = sound_DecodeOggVorbisTrack(pTrack, fileHandle);

	PHYSFS_close(fileHandle);
	return pTrack;
}

void sound_FreeTrack( TRACK *psTrack )
{
#ifndef WZ_NOSOUND
	alDeleteBuffers(1, &psTrack->iBufferName);
	sound_GetError();
#endif
}

#ifndef WZ_NOSOUND
static void sound_AddActiveSample( AUDIO_SAMPLE *psSample )
{
	SAMPLE_LIST *tmp = (SAMPLE_LIST *) malloc( sizeof(SAMPLE_LIST) );

	// Prepend the given sample to our list of active samples
	tmp->curr = psSample;
	tmp->next = active_samples;
	active_samples = tmp;
}

//*
// =======================================================================================================================
// =======================================================================================================================
//
static bool sound_SetupChannel( AUDIO_SAMPLE *psSample )
{
	sound_AddActiveSample( psSample );

	return sound_TrackLooped(psSample->iTrack);
}
#endif

//*
// =======================================================================================================================
// =======================================================================================================================
//
BOOL sound_Play2DSample( TRACK *psTrack, AUDIO_SAMPLE *psSample, BOOL bQueued )
{
#ifndef WZ_NOSOUND
	ALfloat zero[3] = { 0.0, 0.0, 0.0 };
	ALfloat volume;

	if (sfx_volume == 0.0)
	{
		return FALSE;
	}
	volume = ((float)psTrack->iVol / 100.0f);       // each object can have OWN volume!
	psSample->fVol = volume;                        // save computed volume
	volume *= sfx_volume;                           // and now take into account the Users sound Prefs.
	alGenSources( 1, &(psSample->iSample) );
	sound_GetError();
	alSourcef( psSample->iSample, AL_PITCH, 1.0f );
	alSourcef( psSample->iSample, AL_GAIN,volume );
	alSourcefv( psSample->iSample, AL_POSITION, zero );
	alSourcefv( psSample->iSample, AL_VELOCITY, zero );
	alSourcei( psSample->iSample, AL_BUFFER, psTrack->iBufferName );
	alSourcei( psSample->iSample, AL_SOURCE_RELATIVE, AL_TRUE );
	alSourcei( psSample->iSample, AL_LOOPING, (sound_SetupChannel(psSample)) ? AL_TRUE : AL_FALSE );
	sound_GetError();
	alSourcePlay( psSample->iSample );
	sound_GetError();
	if ( bQueued )
	{
		current_queue_sample = psSample->iSample;
	}
	else if ( current_queue_sample == psSample->iSample )
	{
		current_queue_sample = -1;
	}
#endif

	return TRUE;
}

//*
// =======================================================================================================================
// =======================================================================================================================
//
BOOL sound_Play3DSample( TRACK *psTrack, AUDIO_SAMPLE *psSample )
{
#ifndef WZ_NOSOUND
	ALfloat zero[3] = { 0.0, 0.0, 0.0 };
	ALfloat volume;

	if (sfx3d_volume == 0.0)
	{
		return FALSE;
	}

	volume = ((float)psTrack->iVol / 100.0);        // max range is 0-100
	psSample->fVol = volume;                        // store results for later
	alGenSources( 1, &(psSample->iSample) );
	sound_GetError();
	// HACK: this is a workaround for a bug in the 64bit implementation of OpenAL on GNU/Linux
	// The AL_PITCH value really should be 1.0.
	alSourcef( psSample->iSample, AL_PITCH, 1.001 );
	
	sound_SetObjectPosition( psSample );
	alSourcefv( psSample->iSample, AL_VELOCITY, zero );
	alSourcei( psSample->iSample, AL_BUFFER, psTrack->iBufferName );
	alSourcei( psSample->iSample, AL_LOOPING, (sound_SetupChannel(psSample)) ? AL_TRUE : AL_FALSE );
	sound_GetError();
	alSourcePlay( psSample->iSample );
	sound_GetError();
#endif
	return TRUE;
}

//*
// =======================================================================================================================
// =======================================================================================================================
//
BOOL sound_PlayStream( AUDIO_SAMPLE *psSample, const char szFileName[], SDWORD iVol )
{
	return TRUE;
}

//*
// =======================================================================================================================
// =======================================================================================================================
//
void sound_StopSample(AUDIO_SAMPLE* psSample)
{
#ifndef WZ_NOSOUND
	if (psSample->iSample == (ALuint)SAMPLE_NOT_ALLOCATED)
	{
		debug(LOG_SOUND, "sound_StopSample: sample number (%u) out of range, we probably have run out of available OpenAL sources", psSample->iSample);
		return;
	}

	// Tell OpenAL to stop playing the given sample
	alSourceStop(psSample->iSample);
	sound_GetError();
#endif
}

//*
// =======================================================================================================================
// =======================================================================================================================
//
void sound_SetPlayerPos( SDWORD iX, SDWORD iY, SDWORD iZ )
{
#ifndef WZ_NOSOUND
	alListener3f( AL_POSITION, iX, iY, iZ );
	sound_GetError();
#endif
}

//
// =======================================================================================================================
// =======================================================================================================================
//
/** sets player's sound orientation
 * \param iX pitch in degree (current function implementation ignores this)
 * \param iY roll in degree (current function implementation ignores this)
 * \param iZ yaw in degree
 */
void sound_SetPlayerOrientation( SDWORD iX, SDWORD iY, SDWORD iZ )
{
#ifndef WZ_NOSOUND
	//~~~~~~~~~~~
	float	ori[6];
	//~~~~~~~~~~~

	// convert params to rad
	// float pitch = (float)iX * M_PI / 180;
	// float roll = (float)iY * M_PI / 180;
	float yaw = (float)iZ * M_PI / 180;

	ori[0] = -sin( yaw );
	ori[1] = cos( yaw );
	ori[2] = 0;
	ori[3] = 0;
	ori[4] = 0;
	ori[5] = 1;
	alListenerfv( AL_ORIENTATION, ori );
	sound_GetError();
#endif
}

//*
// =======================================================================================================================
// =======================================================================================================================
//
void sound_SetObjectPosition(AUDIO_SAMPLE *psSample)
{
#ifndef WZ_NOSOUND
	//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
	// coordinates
	float	listenerX, listenerY, listenerZ, dX, dY, dZ;

	// calculation results
	float	distance, gain;
	//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

	// compute distance
	alGetListener3f( AL_POSITION, &listenerX, &listenerY, &listenerZ );
	sound_GetError();
	dX = psSample->x  - listenerX; // distances on all axis
	dY = psSample->y - listenerY;
	dZ = psSample->z - listenerZ;
	distance = sqrtf(dX * dX + dY * dY + dZ * dZ); // Pythagorean theorem

	// compute gain         
	gain = (1.0 - (distance * ATTENUATION_FACTOR)) * psSample->fVol * sfx3d_volume;
	if (gain > 1.0)
	{
		gain = 1.0;
	}
	if ( gain < 0.0 )
	{
		gain = 0.0;
	}
	alSourcef( psSample->iSample, AL_GAIN, gain );

	// the alSource3i variant would be better, if it wouldn't provide linker errors however
	alSource3f( psSample->iSample, AL_POSITION, (float)psSample->x,(float)psSample->x,(float)psSample->x );
	sound_GetError();
#endif
}

//*
// =======================================================================================================================
// =======================================================================================================================
//
void sound_PauseSample( AUDIO_SAMPLE *psSample )
{
#ifndef WZ_NOSOUND
	alSourcePause( psSample->iSample );
	sound_GetError();
#endif
}

//*
// =======================================================================================================================
// =======================================================================================================================
//
void sound_ResumeSample( AUDIO_SAMPLE *psSample )
{
#ifndef WZ_NOSOUND
	alSourcePlay( psSample->iSample );
	sound_GetError();
#endif
}

//*
// =======================================================================================================================
// =======================================================================================================================
//
void sound_PauseAll( void )
{
}

//*
// =======================================================================================================================
// =======================================================================================================================
//
void sound_ResumeAll( void )
{
}

//*
// =======================================================================================================================
// =======================================================================================================================
//
void sound_StopAll( void )
{
}

//*
// =======================================================================================================================
// =======================================================================================================================
//
BOOL sound_SampleIsFinished( AUDIO_SAMPLE *psSample )
{
#ifndef WZ_NOSOUND
	//~~~~~~~~~~
	ALenum	state;
	//~~~~~~~~~~

	alGetSourcei( psSample->iSample, AL_SOURCE_STATE, &state );
	sound_GetError(); // check for an error and clear the error state for later on in this function
	if (state == AL_PLAYING || state == AL_PAUSED)
	{
		return FALSE;
	}
	
	if (psSample->iSample != (ALuint)AL_INVALID)
	{
		alDeleteSources(1, &(psSample->iSample));
		sound_GetError();
		psSample->iSample = AL_INVALID;
	}
#endif
	return TRUE;
}

//*
// =======================================================================================================================
// =======================================================================================================================
//

float sound_GetUIVolume()
{
#ifndef WZ_NOSOUND
	return sfx_volume;
#else
	return 0;
#endif
}

void sound_SetUIVolume(float volume)
{
#ifndef WZ_NOSOUND
	sfx_volume = volume;

    // Keep volume in the range of 0.0 - 1.0
	if (sfx_volume < 0.0)
	{
		sfx_volume = 0.0;
	}
	else if (sfx_volume > 1.0)
	{
		sfx_volume = 1.0;
	}
#endif
}

float sound_GetEffectsVolume()
{
#ifndef WZ_NOSOUND
	return sfx3d_volume;
#else
	return 0;
#endif
}

void sound_SetEffectsVolume(float volume)
{
#ifndef WZ_NOSOUND
	sfx3d_volume = volume;

    // Keep volume in the range of 0.0 - 1.0
	if (sfx3d_volume < 0.0)
	{
		sfx3d_volume = 0.0;
	}
	else if (sfx3d_volume > 1.0)
	{
		sfx3d_volume = 1.0;
	}
#endif
}