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doom3-bfg/neo/sound/OpenAL/AL_SoundSample.cpp
Robert Beckebans 699e0e39f5 Fixed problems in the ADPCM decoder. Sound is working.
2013-01-05 19:00:22 +01:00

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/*
===========================================================================
Doom 3 BFG Edition GPL Source Code
Copyright (C) 1993-2012 id Software LLC, a ZeniMax Media company.
Copyright (C) 2013 Robert Beckebans
Copyright (C) 1997-2012 Sam Lantinga <slouken@libsdl.org> (MS ADPCM decoder)
Copyright (c) 2011 Chris Robinson <chris.kcat@gmail.com> (OpenAL helpers)
This file is part of the Doom 3 BFG Edition GPL Source Code ("Doom 3 BFG Edition Source Code").
Doom 3 BFG Edition Source Code 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 3 of the License, or
(at your option) any later version.
Doom 3 BFG Edition Source Code 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 Doom 3 BFG Edition Source Code. If not, see <http://www.gnu.org/licenses/>.
In addition, the Doom 3 BFG Edition Source Code is also subject to certain additional terms. You should have received a copy of these additional terms immediately following the terms and conditions of the GNU General Public License which accompanied the Doom 3 BFG Edition Source Code. If not, please request a copy in writing from id Software at the address below.
If you have questions concerning this license or the applicable additional terms, you may contact in writing id Software LLC, c/o ZeniMax Media Inc., Suite 120, Rockville, Maryland 20850 USA.
===========================================================================
*/
#pragma hdrstop
#include "precompiled.h"
#include "../snd_local.h"
extern idCVar s_useCompression;
extern idCVar s_noSound;
#define GPU_CONVERT_CPU_TO_CPU_CACHED_READONLY_ADDRESS( x ) x
const uint32 SOUND_MAGIC_IDMSA = 0x6D7A7274;
extern idCVar sys_lang;
/*
========================
AllocBuffer
========================
*/
static void* AllocBuffer( int size, const char* name )
{
return Mem_Alloc( size, TAG_AUDIO );
}
/*
========================
FreeBuffer
========================
*/
static void FreeBuffer( void* p )
{
return Mem_Free( p );
}
/*
========================
idSoundSample_OpenAL::idSoundSample_OpenAL
========================
*/
idSoundSample_OpenAL::idSoundSample_OpenAL()
{
timestamp = FILE_NOT_FOUND_TIMESTAMP;
loaded = false;
neverPurge = false;
levelLoadReferenced = false;
memset( &format, 0, sizeof( format ) );
totalBufferSize = 0;
playBegin = 0;
playLength = 0;
lastPlayedTime = 0;
openalBuffer = 0;
}
/*
========================
idSoundSample_OpenAL::~idSoundSample_OpenAL
========================
*/
idSoundSample_OpenAL::~idSoundSample_OpenAL()
{
FreeData();
}
/*
========================
idSoundSample_OpenAL::WriteGeneratedSample
========================
*/
void idSoundSample_OpenAL::WriteGeneratedSample( idFile* fileOut )
{
fileOut->WriteBig( SOUND_MAGIC_IDMSA );
fileOut->WriteBig( timestamp );
fileOut->WriteBig( loaded );
fileOut->WriteBig( playBegin );
fileOut->WriteBig( playLength );
idWaveFile::WriteWaveFormatDirect( format, fileOut );
fileOut->WriteBig( ( int )amplitude.Num() );
fileOut->Write( amplitude.Ptr(), amplitude.Num() );
fileOut->WriteBig( totalBufferSize );
fileOut->WriteBig( ( int )buffers.Num() );
for( int i = 0; i < buffers.Num(); i++ )
{
fileOut->WriteBig( buffers[ i ].numSamples );
fileOut->WriteBig( buffers[ i ].bufferSize );
fileOut->Write( buffers[ i ].buffer, buffers[ i ].bufferSize );
};
}
/*
========================
idSoundSample_OpenAL::WriteAllSamples
========================
*/
void idSoundSample_OpenAL::WriteAllSamples( const idStr& sampleName )
{
idSoundSample_OpenAL* samplePC = new idSoundSample_OpenAL();
{
idStrStatic< MAX_OSPATH > inName = sampleName;
inName.Append( ".msadpcm" );
idStrStatic< MAX_OSPATH > inName2 = sampleName;
inName2.Append( ".wav" );
idStrStatic< MAX_OSPATH > outName = "generated/";
outName.Append( sampleName );
outName.Append( ".idwav" );
if( samplePC->LoadWav( inName ) || samplePC->LoadWav( inName2 ) )
{
idFile* fileOut = fileSystem->OpenFileWrite( outName, "fs_basepath" );
samplePC->WriteGeneratedSample( fileOut );
delete fileOut;
}
}
delete samplePC;
}
/*
========================
idSoundSample_OpenAL::LoadGeneratedSound
========================
*/
bool idSoundSample_OpenAL::LoadGeneratedSample( const idStr& filename )
{
#if 1
idFileLocal fileIn( fileSystem->OpenFileReadMemory( filename ) );
if( fileIn != NULL )
{
uint32 magic;
fileIn->ReadBig( magic );
fileIn->ReadBig( timestamp );
fileIn->ReadBig( loaded );
fileIn->ReadBig( playBegin );
fileIn->ReadBig( playLength );
idWaveFile::ReadWaveFormatDirect( format, fileIn );
int num;
fileIn->ReadBig( num );
amplitude.Clear();
amplitude.SetNum( num );
fileIn->Read( amplitude.Ptr(), amplitude.Num() );
fileIn->ReadBig( totalBufferSize );
fileIn->ReadBig( num );
buffers.SetNum( num );
for( int i = 0; i < num; i++ )
{
fileIn->ReadBig( buffers[ i ].numSamples );
fileIn->ReadBig( buffers[ i ].bufferSize );
buffers[ i ].buffer = AllocBuffer( buffers[ i ].bufferSize, GetName() );
fileIn->Read( buffers[ i ].buffer, buffers[ i ].bufferSize );
buffers[ i ].buffer = GPU_CONVERT_CPU_TO_CPU_CACHED_READONLY_ADDRESS( buffers[ i ].buffer );
}
return true;
}
#endif
return false;
}
/*
========================
idSoundSample_OpenAL::Load
========================
*/
void idSoundSample_OpenAL::LoadResource()
{
FreeData();
if( idStr::Icmpn( GetName(), "_default", 8 ) == 0 )
{
MakeDefault();
return;
}
if( s_noSound.GetBool() )
{
MakeDefault();
return;
}
loaded = false;
for( int i = 0; i < 2; i++ )
{
idStrStatic< MAX_OSPATH > sampleName = GetName();
if( ( i == 0 ) && !sampleName.Replace( "/vo/", va( "/vo/%s/", sys_lang.GetString() ) ) )
{
i++;
}
idStrStatic< MAX_OSPATH > generatedName = "generated/";
generatedName.Append( sampleName );
{
if( s_useCompression.GetBool() )
{
sampleName.Append( ".msadpcm" );
}
else
{
sampleName.Append( ".wav" );
}
generatedName.Append( ".idwav" );
}
loaded = LoadGeneratedSample( generatedName ) || LoadWav( sampleName );
if( !loaded && s_useCompression.GetBool() )
{
sampleName.SetFileExtension( "wav" );
loaded = LoadWav( sampleName );
}
if( loaded )
{
if( cvarSystem->GetCVarBool( "fs_buildresources" ) )
{
fileSystem->AddSamplePreload( GetName() );
WriteAllSamples( GetName() );
if( sampleName.Find( "/vo/" ) >= 0 )
{
for( int i = 0; i < Sys_NumLangs(); i++ )
{
const char* lang = Sys_Lang( i );
if( idStr::Icmp( lang, ID_LANG_ENGLISH ) == 0 )
{
continue;
}
idStrStatic< MAX_OSPATH > locName = GetName();
locName.Replace( "/vo/", va( "/vo/%s/", Sys_Lang( i ) ) );
WriteAllSamples( locName );
}
}
}
// upload PCM data to OpenAL
CreateOpenALBuffer();
return;
}
}
if( !loaded )
{
// make it default if everything else fails
MakeDefault();
}
return;
}
void idSoundSample_OpenAL::CreateOpenALBuffer()
{
// build OpenAL buffer
alGetError();
alGenBuffers( 1, &openalBuffer );
if( alGetError() != AL_NO_ERROR )
{
common->Error( "idSoundSample_OpenAL::CreateOpenALBuffer: error generating OpenAL hardware buffer" );
}
if( alIsBuffer( openalBuffer ) )
{
alGetError();
// RB: TODO decode idWaveFile::FORMAT_ADPCM to idWaveFile::FORMAT_PCM
// and build one big OpenAL buffer using the alBufferSubData extension
void* buffer = NULL;
uint32 bufferSize = 0;
if( format.basic.formatTag == idWaveFile::FORMAT_ADPCM )
{
buffer = buffers[0].buffer;
bufferSize = buffers[0].bufferSize;
if( MS_ADPCM_decode( ( uint8** ) &buffer, &bufferSize ) < 0 )
{
common->Error( "idSoundSample_OpenAL::CreateOpenALBuffer: could not decode ADPCM '%s' to 16 bit format", GetName() );
}
buffers[0].buffer = buffer;
buffers[0].bufferSize = bufferSize;
totalBufferSize = bufferSize;
}
else if( format.basic.formatTag == idWaveFile::FORMAT_XMA2 )
{
common->Error( "idSoundSample_OpenAL::CreateOpenALBuffer: could not decode XMA2 '%s' to 16 bit format", GetName() );
}
else if( format.basic.formatTag == idWaveFile::FORMAT_EXTENSIBLE )
{
common->Error( "idSoundSample_OpenAL::CreateOpenALBuffer: could not decode extensible WAV format '%s' to 16 bit format", GetName() );
}
else
{
// TODO concatenate buffers
assert( buffers.Num() == 1 );
buffer = buffers[0].buffer;
bufferSize = buffers[0].bufferSize;
}
#if 0
if( alIsExtensionPresent( "AL_SOFT_buffer_samples" ) )
{
ALenum type = AL_SHORT_SOFT;
if( format.basic.bitsPerSample != 16 )
{
//common->Error( "idSoundSample_OpenAL::LoadResource: '%s' not a 16 bit format", GetName() );
}
ALenum channels = NumChannels() == 1 ? AL_MONO_SOFT : AL_STEREO_SOFT;
ALenum alFormat = GetOpenALSoftFormat( channels, type );
alBufferSamplesSOFT( openalBuffer, format.basic.samplesPerSec, alFormat, BytesToFrames( bufferSize, channels, type ), channels, type, buffer );
}
else
#endif
{
alBufferData( openalBuffer, GetOpenALBufferFormat(), buffer, bufferSize, format.basic.samplesPerSec );
}
if( alGetError() != AL_NO_ERROR )
{
common->Error( "idSoundSample_OpenAL::CreateOpenALBuffer: error loading data into OpenAL hardware buffer" );
}
}
}
/*
========================
idSoundSample_OpenAL::LoadWav
========================
*/
bool idSoundSample_OpenAL::LoadWav( const idStr& filename )
{
// load the wave
idWaveFile wave;
if( !wave.Open( filename ) )
{
return false;
}
idStrStatic< MAX_OSPATH > sampleName = filename;
sampleName.SetFileExtension( "amp" );
LoadAmplitude( sampleName );
const char* formatError = wave.ReadWaveFormat( format );
if( formatError != NULL )
{
idLib::Warning( "LoadWav( %s ) : %s", filename.c_str(), formatError );
MakeDefault();
return false;
}
timestamp = wave.Timestamp();
totalBufferSize = wave.SeekToChunk( 'data' );
if( format.basic.formatTag == idWaveFile::FORMAT_PCM || format.basic.formatTag == idWaveFile::FORMAT_EXTENSIBLE )
{
if( format.basic.bitsPerSample != 16 )
{
idLib::Warning( "LoadWav( %s ) : %s", filename.c_str(), "Not a 16 bit PCM wav file" );
MakeDefault();
return false;
}
playBegin = 0;
playLength = ( totalBufferSize ) / format.basic.blockSize;
buffers.SetNum( 1 );
buffers[0].bufferSize = totalBufferSize;
buffers[0].numSamples = playLength;
buffers[0].buffer = AllocBuffer( totalBufferSize, GetName() );
wave.Read( buffers[0].buffer, totalBufferSize );
if( format.basic.bitsPerSample == 16 )
{
idSwap::LittleArray( ( short* )buffers[0].buffer, totalBufferSize / sizeof( short ) );
}
buffers[0].buffer = GPU_CONVERT_CPU_TO_CPU_CACHED_READONLY_ADDRESS( buffers[0].buffer );
}
else if( format.basic.formatTag == idWaveFile::FORMAT_ADPCM )
{
playBegin = 0;
playLength = ( ( totalBufferSize / format.basic.blockSize ) * format.extra.adpcm.samplesPerBlock );
buffers.SetNum( 1 );
buffers[0].bufferSize = totalBufferSize;
buffers[0].numSamples = playLength;
buffers[0].buffer = AllocBuffer( totalBufferSize, GetName() );
wave.Read( buffers[0].buffer, totalBufferSize );
buffers[0].buffer = GPU_CONVERT_CPU_TO_CPU_CACHED_READONLY_ADDRESS( buffers[0].buffer );
}
else if( format.basic.formatTag == idWaveFile::FORMAT_XMA2 )
{
if( format.extra.xma2.blockCount == 0 )
{
idLib::Warning( "LoadWav( %s ) : %s", filename.c_str(), "No data blocks in file" );
MakeDefault();
return false;
}
int bytesPerBlock = format.extra.xma2.bytesPerBlock;
assert( format.extra.xma2.blockCount == ALIGN( totalBufferSize, bytesPerBlock ) / bytesPerBlock );
assert( format.extra.xma2.blockCount * bytesPerBlock >= totalBufferSize );
assert( format.extra.xma2.blockCount * bytesPerBlock < totalBufferSize + bytesPerBlock );
buffers.SetNum( format.extra.xma2.blockCount );
for( int i = 0; i < buffers.Num(); i++ )
{
if( i == buffers.Num() - 1 )
{
buffers[i].bufferSize = totalBufferSize - ( i * bytesPerBlock );
}
else
{
buffers[i].bufferSize = bytesPerBlock;
}
buffers[i].buffer = AllocBuffer( buffers[i].bufferSize, GetName() );
wave.Read( buffers[i].buffer, buffers[i].bufferSize );
buffers[i].buffer = GPU_CONVERT_CPU_TO_CPU_CACHED_READONLY_ADDRESS( buffers[i].buffer );
}
int seekTableSize = wave.SeekToChunk( 'seek' );
if( seekTableSize != 4 * buffers.Num() )
{
idLib::Warning( "LoadWav( %s ) : %s", filename.c_str(), "Wrong number of entries in seek table" );
MakeDefault();
return false;
}
for( int i = 0; i < buffers.Num(); i++ )
{
wave.Read( &buffers[i].numSamples, sizeof( buffers[i].numSamples ) );
idSwap::Big( buffers[i].numSamples );
}
playBegin = format.extra.xma2.loopBegin;
playLength = format.extra.xma2.loopLength;
if( buffers[buffers.Num() - 1].numSamples < playBegin + playLength )
{
// This shouldn't happen, but it's not fatal if it does
playLength = buffers[buffers.Num() - 1].numSamples - playBegin;
}
else
{
// Discard samples beyond playLength
for( int i = 0; i < buffers.Num(); i++ )
{
if( buffers[i].numSamples > playBegin + playLength )
{
buffers[i].numSamples = playBegin + playLength;
// Ideally, the following loop should always have 0 iterations because playBegin + playLength ends in the last block already
// But there is no guarantee for that, so to be safe, discard all buffers beyond this one
for( int j = i + 1; j < buffers.Num(); j++ )
{
FreeBuffer( buffers[j].buffer );
}
buffers.SetNum( i + 1 );
break;
}
}
}
}
else
{
idLib::Warning( "LoadWav( %s ) : Unsupported wave format %d", filename.c_str(), format.basic.formatTag );
MakeDefault();
return false;
}
wave.Close();
if( format.basic.formatTag == idWaveFile::FORMAT_EXTENSIBLE )
{
// HACK: XAudio2 doesn't really support FORMAT_EXTENSIBLE so we convert it to a basic format after extracting the channel mask
format.basic.formatTag = format.extra.extensible.subFormat.data1;
}
// sanity check...
assert( buffers[buffers.Num() - 1].numSamples == playBegin + playLength );
return true;
}
/*
========================
idSoundSample_OpenAL::MakeDefault
========================
*/
void idSoundSample_OpenAL::MakeDefault()
{
FreeData();
static const int DEFAULT_NUM_SAMPLES = 4096;
timestamp = FILE_NOT_FOUND_TIMESTAMP;
loaded = true;
memset( &format, 0, sizeof( format ) );
format.basic.formatTag = idWaveFile::FORMAT_PCM;
format.basic.numChannels = 1;
format.basic.bitsPerSample = 16;
format.basic.samplesPerSec = 22050; //44100; //XAUDIO2_MIN_SAMPLE_RATE;
format.basic.blockSize = format.basic.numChannels * format.basic.bitsPerSample / 8;
format.basic.avgBytesPerSec = format.basic.samplesPerSec * format.basic.blockSize;
assert( format.basic.blockSize == 2 );
totalBufferSize = DEFAULT_NUM_SAMPLES * 2;// * sizeof( short );
short* defaultBuffer = ( short* )AllocBuffer( totalBufferSize, GetName() );
for( int i = 0; i < DEFAULT_NUM_SAMPLES; i += 2 )
{
float v = sin( idMath::PI * 2 * i / 64 );
int sample = v * 0x4000;
defaultBuffer[i + 0] = sample;
defaultBuffer[i + 1] = sample;
//defaultBuffer[i + 0] = SHRT_MIN;
//defaultBuffer[i + 1] = SHRT_MAX;
}
buffers.SetNum( 1 );
buffers[0].buffer = defaultBuffer;
buffers[0].bufferSize = totalBufferSize;
buffers[0].numSamples = DEFAULT_NUM_SAMPLES;
buffers[0].buffer = GPU_CONVERT_CPU_TO_CPU_CACHED_READONLY_ADDRESS( buffers[0].buffer );
playBegin = 0;
playLength = DEFAULT_NUM_SAMPLES;
alGetError();
alGenBuffers( 1, &openalBuffer );
if( alGetError() != AL_NO_ERROR )
{
common->Error( "idSoundSample_OpenAL::MakeDefault: error generating OpenAL hardware buffer" );
}
if( alIsBuffer( openalBuffer ) )
{
alGetError();
alBufferData( openalBuffer, GetOpenALBufferFormat(), defaultBuffer, totalBufferSize, format.basic.samplesPerSec );
if( alGetError() != AL_NO_ERROR )
{
common->Error( "idSoundSample_OpenAL::MakeDefault: error loading data into OpenAL hardware buffer" );
}
}
}
/*
========================
idSoundSample_OpenAL::FreeData
Called before deleting the object and at the start of LoadResource()
========================
*/
void idSoundSample_OpenAL::FreeData()
{
if( buffers.Num() > 0 )
{
soundSystemLocal.StopVoicesWithSample( ( idSoundSample* )this );
for( int i = 0; i < buffers.Num(); i++ )
{
FreeBuffer( buffers[i].buffer );
}
buffers.Clear();
}
amplitude.Clear();
timestamp = FILE_NOT_FOUND_TIMESTAMP;
memset( &format, 0, sizeof( format ) );
loaded = false;
totalBufferSize = 0;
playBegin = 0;
playLength = 0;
if( alIsBuffer( openalBuffer ) )
{
alGetError();
alDeleteBuffers( 1, &openalBuffer );
if( alGetError() != AL_NO_ERROR )
{
common->Error( "idSoundSample_OpenAL::FreeData: error unloading data from OpenAL hardware buffer" );
}
else
{
openalBuffer = 0;
}
}
}
/*
========================
idSoundSample_OpenAL::LoadAmplitude
========================
*/
bool idSoundSample_OpenAL::LoadAmplitude( const idStr& name )
{
amplitude.Clear();
idFileLocal f( fileSystem->OpenFileRead( name ) );
if( f == NULL )
{
return false;
}
amplitude.SetNum( f->Length() );
f->Read( amplitude.Ptr(), amplitude.Num() );
return true;
}
/*
========================
idSoundSample_OpenAL::GetAmplitude
========================
*/
float idSoundSample_OpenAL::GetAmplitude( int timeMS ) const
{
if( timeMS < 0 || timeMS > LengthInMsec() )
{
return 0.0f;
}
if( IsDefault() )
{
return 1.0f;
}
int index = timeMS * 60 / 1000;
if( index < 0 || index >= amplitude.Num() )
{
return 0.0f;
}
return ( float )amplitude[index] / 255.0f;
}
const char* idSoundSample_OpenAL::OpenALSoftChannelsName( ALenum chans ) const
{
switch( chans )
{
case AL_MONO_SOFT:
return "Mono";
case AL_STEREO_SOFT:
return "Stereo";
case AL_REAR_SOFT:
return "Rear";
case AL_QUAD_SOFT:
return "Quadraphonic";
case AL_5POINT1_SOFT:
return "5.1 Surround";
case AL_6POINT1_SOFT:
return "6.1 Surround";
case AL_7POINT1_SOFT:
return "7.1 Surround";
}
return "Unknown Channels";
}
const char* idSoundSample_OpenAL::OpenALSoftTypeName( ALenum type ) const
{
switch( type )
{
case AL_BYTE_SOFT:
return "S8";
case AL_UNSIGNED_BYTE_SOFT:
return "U8";
case AL_SHORT_SOFT:
return "S16";
case AL_UNSIGNED_SHORT_SOFT:
return "U16";
case AL_INT_SOFT:
return "S32";
case AL_UNSIGNED_INT_SOFT:
return "U32";
case AL_FLOAT_SOFT:
return "Float32";
case AL_DOUBLE_SOFT:
return "Float64";
}
return "Unknown Type";
}
ALsizei idSoundSample_OpenAL::FramesToBytes( ALsizei size, ALenum channels, ALenum type ) const
{
switch( channels )
{
case AL_MONO_SOFT:
size *= 1;
break;
case AL_STEREO_SOFT:
size *= 2;
break;
case AL_REAR_SOFT:
size *= 2;
break;
case AL_QUAD_SOFT:
size *= 4;
break;
case AL_5POINT1_SOFT:
size *= 6;
break;
case AL_6POINT1_SOFT:
size *= 7;
break;
case AL_7POINT1_SOFT:
size *= 8;
break;
}
switch( type )
{
case AL_BYTE_SOFT:
size *= sizeof( ALbyte );
break;
case AL_UNSIGNED_BYTE_SOFT:
size *= sizeof( ALubyte );
break;
case AL_SHORT_SOFT:
size *= sizeof( ALshort );
break;
case AL_UNSIGNED_SHORT_SOFT:
size *= sizeof( ALushort );
break;
case AL_INT_SOFT:
size *= sizeof( ALint );
break;
case AL_UNSIGNED_INT_SOFT:
size *= sizeof( ALuint );
break;
case AL_FLOAT_SOFT:
size *= sizeof( ALfloat );
break;
case AL_DOUBLE_SOFT:
size *= sizeof( ALdouble );
break;
}
return size;
}
ALsizei idSoundSample_OpenAL::BytesToFrames( ALsizei size, ALenum channels, ALenum type ) const
{
return size / FramesToBytes( 1, channels, type );
}
ALenum idSoundSample_OpenAL::GetOpenALSoftFormat( ALenum channels, ALenum type ) const
{
ALenum format = AL_NONE;
/* If using AL_SOFT_buffer_samples, try looking through its formats */
if( alIsExtensionPresent( "AL_SOFT_buffer_samples" ) )
{
/* AL_SOFT_buffer_samples is more lenient with matching formats. The
* specified sample type does not need to match the returned format,
* but it is nice to try to get something close. */
if( type == AL_UNSIGNED_BYTE_SOFT || type == AL_BYTE_SOFT )
{
if( channels == AL_MONO_SOFT ) format = AL_MONO8_SOFT;
else if( channels == AL_STEREO_SOFT ) format = AL_STEREO8_SOFT;
else if( channels == AL_QUAD_SOFT ) format = AL_QUAD8_SOFT;
else if( channels == AL_5POINT1_SOFT ) format = AL_5POINT1_8_SOFT;
else if( channels == AL_6POINT1_SOFT ) format = AL_6POINT1_8_SOFT;
else if( channels == AL_7POINT1_SOFT ) format = AL_7POINT1_8_SOFT;
}
else if( type == AL_UNSIGNED_SHORT_SOFT || type == AL_SHORT_SOFT )
{
if( channels == AL_MONO_SOFT ) format = AL_MONO16_SOFT;
else if( channels == AL_STEREO_SOFT ) format = AL_STEREO16_SOFT;
else if( channels == AL_QUAD_SOFT ) format = AL_QUAD16_SOFT;
else if( channels == AL_5POINT1_SOFT ) format = AL_5POINT1_16_SOFT;
else if( channels == AL_6POINT1_SOFT ) format = AL_6POINT1_16_SOFT;
else if( channels == AL_7POINT1_SOFT ) format = AL_7POINT1_16_SOFT;
}
else if( type == AL_UNSIGNED_BYTE3_SOFT || type == AL_BYTE3_SOFT ||
type == AL_UNSIGNED_INT_SOFT || type == AL_INT_SOFT ||
type == AL_FLOAT_SOFT || type == AL_DOUBLE_SOFT )
{
if( channels == AL_MONO_SOFT ) format = AL_MONO32F_SOFT;
else if( channels == AL_STEREO_SOFT ) format = AL_STEREO32F_SOFT;
else if( channels == AL_QUAD_SOFT ) format = AL_QUAD32F_SOFT;
else if( channels == AL_5POINT1_SOFT ) format = AL_5POINT1_32F_SOFT;
else if( channels == AL_6POINT1_SOFT ) format = AL_6POINT1_32F_SOFT;
else if( channels == AL_7POINT1_SOFT ) format = AL_7POINT1_32F_SOFT;
}
if( format != AL_NONE && !alIsBufferFormatSupportedSOFT( format ) )
format = AL_NONE;
/* A matching format was not found or supported. Try 32-bit float. */
if( format == AL_NONE )
{
if( channels == AL_MONO_SOFT ) format = AL_MONO32F_SOFT;
else if( channels == AL_STEREO_SOFT ) format = AL_STEREO32F_SOFT;
else if( channels == AL_QUAD_SOFT ) format = AL_QUAD32F_SOFT;
else if( channels == AL_5POINT1_SOFT ) format = AL_5POINT1_32F_SOFT;
else if( channels == AL_6POINT1_SOFT ) format = AL_6POINT1_32F_SOFT;
else if( channels == AL_7POINT1_SOFT ) format = AL_7POINT1_32F_SOFT;
if( format != AL_NONE && !alIsBufferFormatSupportedSOFT( format ) )
format = AL_NONE;
}
/* 32-bit float not supported. Try 16-bit int. */
if( format == AL_NONE )
{
if( channels == AL_MONO_SOFT ) format = AL_MONO16_SOFT;
else if( channels == AL_STEREO_SOFT ) format = AL_STEREO16_SOFT;
else if( channels == AL_QUAD_SOFT ) format = AL_QUAD16_SOFT;
else if( channels == AL_5POINT1_SOFT ) format = AL_5POINT1_16_SOFT;
else if( channels == AL_6POINT1_SOFT ) format = AL_6POINT1_16_SOFT;
else if( channels == AL_7POINT1_SOFT ) format = AL_7POINT1_16_SOFT;
if( format != AL_NONE && !alIsBufferFormatSupportedSOFT( format ) )
format = AL_NONE;
}
/* 16-bit int not supported. Try 8-bit int. */
if( format == AL_NONE )
{
if( channels == AL_MONO_SOFT ) format = AL_MONO8_SOFT;
else if( channels == AL_STEREO_SOFT ) format = AL_STEREO8_SOFT;
else if( channels == AL_QUAD_SOFT ) format = AL_QUAD8_SOFT;
else if( channels == AL_5POINT1_SOFT ) format = AL_5POINT1_8_SOFT;
else if( channels == AL_6POINT1_SOFT ) format = AL_6POINT1_8_SOFT;
else if( channels == AL_7POINT1_SOFT ) format = AL_7POINT1_8_SOFT;
if( format != AL_NONE && !alIsBufferFormatSupportedSOFT( format ) )
format = AL_NONE;
}
return format;
}
/* We use the AL_EXT_MCFORMATS extension to provide output of Quad, 5.1,
* and 7.1 channel configs, AL_EXT_FLOAT32 for 32-bit float samples, and
* AL_EXT_DOUBLE for 64-bit float samples. */
if( type == AL_UNSIGNED_BYTE_SOFT )
{
if( channels == AL_MONO_SOFT )
format = AL_FORMAT_MONO8;
else if( channels == AL_STEREO_SOFT )
format = AL_FORMAT_STEREO8;
else if( alIsExtensionPresent( "AL_EXT_MCFORMATS" ) )
{
if( channels == AL_QUAD_SOFT )
format = alGetEnumValue( "AL_FORMAT_QUAD8" );
else if( channels == AL_5POINT1_SOFT )
format = alGetEnumValue( "AL_FORMAT_51CHN8" );
else if( channels == AL_6POINT1_SOFT )
format = alGetEnumValue( "AL_FORMAT_61CHN8" );
else if( channels == AL_7POINT1_SOFT )
format = alGetEnumValue( "AL_FORMAT_71CHN8" );
}
}
else if( type == AL_SHORT_SOFT )
{
if( channels == AL_MONO_SOFT )
format = AL_FORMAT_MONO16;
else if( channels == AL_STEREO_SOFT )
format = AL_FORMAT_STEREO16;
else if( alIsExtensionPresent( "AL_EXT_MCFORMATS" ) )
{
if( channels == AL_QUAD_SOFT )
format = alGetEnumValue( "AL_FORMAT_QUAD16" );
else if( channels == AL_5POINT1_SOFT )
format = alGetEnumValue( "AL_FORMAT_51CHN16" );
else if( channels == AL_6POINT1_SOFT )
format = alGetEnumValue( "AL_FORMAT_61CHN16" );
else if( channels == AL_7POINT1_SOFT )
format = alGetEnumValue( "AL_FORMAT_71CHN16" );
}
}
else if( type == AL_FLOAT_SOFT && alIsExtensionPresent( "AL_EXT_FLOAT32" ) )
{
if( channels == AL_MONO_SOFT )
format = alGetEnumValue( "AL_FORMAT_MONO_FLOAT32" );
else if( channels == AL_STEREO_SOFT )
format = alGetEnumValue( "AL_FORMAT_STEREO_FLOAT32" );
else if( alIsExtensionPresent( "AL_EXT_MCFORMATS" ) )
{
if( channels == AL_QUAD_SOFT )
format = alGetEnumValue( "AL_FORMAT_QUAD32" );
else if( channels == AL_5POINT1_SOFT )
format = alGetEnumValue( "AL_FORMAT_51CHN32" );
else if( channels == AL_6POINT1_SOFT )
format = alGetEnumValue( "AL_FORMAT_61CHN32" );
else if( channels == AL_7POINT1_SOFT )
format = alGetEnumValue( "AL_FORMAT_71CHN32" );
}
}
else if( type == AL_DOUBLE_SOFT && alIsExtensionPresent( "AL_EXT_DOUBLE" ) )
{
if( channels == AL_MONO_SOFT )
format = alGetEnumValue( "AL_FORMAT_MONO_DOUBLE" );
else if( channels == AL_STEREO_SOFT )
format = alGetEnumValue( "AL_FORMAT_STEREO_DOUBLE" );
}
/* NOTE: It seems OSX returns -1 from alGetEnumValue for unknown enums, as
* opposed to 0. Correct it. */
if( format == -1 )
format = 0;
return format;
}
ALenum idSoundSample_OpenAL::GetOpenALBufferFormat() const
{
ALenum alFormat;
#if 0
if( alIsExtensionPresent( "AL_SOFT_buffer_samples" ) )
{
if( format.basic.formatTag == idWaveFile::FORMAT_PCM )
{
alFormat = NumChannels() == 1 ? AL_MONO16_SOFT : AL_STEREO16_SOFT;
}
else if( format.basic.formatTag == idWaveFile::FORMAT_ADPCM )
{
alFormat = NumChannels() == 1 ? AL_MONO8_SOFT : AL_STEREO8_SOFT;
//alFormat = NumChannels() == 1 ? AL_MONO16_SOFT : AL_STEREO16_SOFT;
}
else if( format.basic.formatTag == idWaveFile::FORMAT_XMA2 )
{
alFormat = NumChannels() == 1 ? AL_MONO16_SOFT : AL_STEREO16_SOFT;
}
else
{
alFormat = NumChannels() == 1 ? AL_MONO16_SOFT : AL_STEREO16_SOFT;
}
if( !alIsBufferFormatSupportedSOFT( alFormat ) )
{
alFormat = NumChannels() == 1 ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16;
}
}
else
#endif
{
if( format.basic.formatTag == idWaveFile::FORMAT_PCM )
{
alFormat = NumChannels() == 1 ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16;
}
else if( format.basic.formatTag == idWaveFile::FORMAT_ADPCM )
{
//alFormat = NumChannels() == 1 ? AL_FORMAT_MONO8 : AL_FORMAT_STEREO8;
alFormat = NumChannels() == 1 ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16;
//alFormat = NumChannels() == 1 ? AL_FORMAT_MONO_IMA4 : AL_FORMAT_STEREO_IMA4;
}
else if( format.basic.formatTag == idWaveFile::FORMAT_XMA2 )
{
alFormat = NumChannels() == 1 ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16;
}
else
{
alFormat = NumChannels() == 1 ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16;
}
}
return alFormat;
}
int32 idSoundSample_OpenAL::MS_ADPCM_nibble( MS_ADPCM_decodeState_t* state, int8 nybble )
{
const int32 max_audioval = ( ( 1 << ( 16 - 1 ) ) - 1 );
const int32 min_audioval = -( 1 << ( 16 - 1 ) );
const int32 adaptive[] =
{
230, 230, 230, 230, 307, 409, 512, 614,
768, 614, 512, 409, 307, 230, 230, 230
};
int32 new_sample, delta;
new_sample = ( ( state->iSamp1 * state->coef1 ) +
( state->iSamp2 * state->coef2 ) ) / 256;
if( nybble & 0x08 )
{
new_sample += state->iDelta * ( nybble - 0x10 );
}
else
{
new_sample += state->iDelta * nybble;
}
if( new_sample < min_audioval )
{
new_sample = min_audioval;
}
else if( new_sample > max_audioval )
{
new_sample = max_audioval;
}
delta = ( ( int32 ) state->iDelta * adaptive[nybble] ) / 256;
if( delta < 16 )
{
delta = 16;
}
state->iDelta = ( uint16 ) delta;
state->iSamp2 = state->iSamp1;
state->iSamp1 = ( int16 ) new_sample;
return ( new_sample );
}
int idSoundSample_OpenAL::MS_ADPCM_decode( uint8** audio_buf, uint32* audio_len )
{
static MS_ADPCM_decodeState_t states[2];
MS_ADPCM_decodeState_t* state[2];
uint8* freeable, *encoded, *decoded;
int32 encoded_len, samplesleft;
int8 nybble;
int8 stereo;
int32 new_sample;
// Allocate the proper sized output buffer
encoded_len = *audio_len;
encoded = *audio_buf;
freeable = *audio_buf;
*audio_len = ( encoded_len / format.basic.blockSize ) * format.extra.adpcm.samplesPerBlock * format.basic.numChannels * sizeof( int16 );
*audio_buf = ( uint8* ) Mem_Alloc( *audio_len, TAG_AUDIO );
if( *audio_buf == NULL )
{
//SDL_Error( SDL_ENOMEM );
return ( -1 );
}
decoded = *audio_buf;
assert( format.basic.numChannels == 1 || format.basic.numChannels == 2 );
// Get ready... Go!
stereo = ( format.basic.numChannels == 2 ) ? 1 : 0;
state[0] = &states[0];
state[1] = &states[stereo];
while( encoded_len >= format.basic.blockSize )
{
// Grab the initial information for this block
state[0]->hPredictor = *encoded++;
assert( state[0]->hPredictor < format.extra.adpcm.numCoef );
state[0]->hPredictor = idMath::ClampInt( 0, 6, state[0]->hPredictor );
state[0]->coef1 = format.extra.adpcm.aCoef[state[0]->hPredictor].coef1;
state[0]->coef2 = format.extra.adpcm.aCoef[state[0]->hPredictor].coef2;
if( stereo )
{
state[1]->hPredictor = *encoded++;
assert( state[1]->hPredictor < format.extra.adpcm.numCoef );
state[1]->hPredictor = idMath::ClampInt( 0, 6, state[1]->hPredictor );
state[1]->coef1 = format.extra.adpcm.aCoef[state[1]->hPredictor].coef1;
state[1]->coef2 = format.extra.adpcm.aCoef[state[1]->hPredictor].coef2;
}
state[0]->iDelta = ( ( encoded[1] << 8 ) | encoded[0] );
encoded += sizeof( int16 );
if( stereo )
{
state[1]->iDelta = ( ( encoded[1] << 8 ) | encoded[0] );
encoded += sizeof( int16 );
}
state[0]->iSamp1 = ( ( encoded[1] << 8 ) | encoded[0] );
encoded += sizeof( int16 );
if( stereo )
{
state[1]->iSamp1 = ( ( encoded[1] << 8 ) | encoded[0] );
encoded += sizeof( int16 );
}
state[0]->iSamp2 = ( ( encoded[1] << 8 ) | encoded[0] );
encoded += sizeof( int16 );
if( stereo )
{
state[1]->iSamp2 = ( ( encoded[1] << 8 ) | encoded[0] );
encoded += sizeof( int16 );
}
// Store the two initial samples we start with
decoded[0] = state[0]->iSamp2 & 0xFF;
decoded[1] = ( state[0]->iSamp2 >> 8 ) & 0xFF;
decoded += 2;
if( stereo )
{
decoded[0] = state[1]->iSamp2 & 0xFF;
decoded[1] = ( state[1]->iSamp2 >> 8 ) & 0xFF;
decoded += 2;
}
decoded[0] = state[0]->iSamp1 & 0xFF;
decoded[1] = ( state[0]->iSamp1 >> 8 ) & 0xFF;
decoded += 2;
if( stereo )
{
decoded[0] = state[1]->iSamp1 & 0xFF;
decoded[1] = ( state[1]->iSamp1 >> 8 ) & 0xFF;
decoded += 2;
}
// Decode and store the other samples in this block
samplesleft = ( format.extra.adpcm.samplesPerBlock - 2 ) * format.basic.numChannels;
while( samplesleft > 0 )
{
nybble = ( *encoded ) >> 4;
new_sample = MS_ADPCM_nibble( state[0], nybble );
decoded[0] = new_sample & 0xFF;
decoded[1] = ( new_sample >> 8 ) & 0xFF;
decoded += 2;
nybble = ( *encoded ) & 0x0F;
new_sample = MS_ADPCM_nibble( state[1], nybble );
decoded[0] = new_sample & 0xFF;
decoded[1] = ( new_sample >> 8 ) & 0xFF;
decoded += 2;
++encoded;
samplesleft -= 2;
}
encoded_len -= format.basic.blockSize;
}
Mem_Free( freeable );
return 0;
}
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