mirror of
https://github.com/id-Software/DOOM-3-BFG.git
synced 2024-12-02 08:51:57 +00:00
Added MS ADPCM decoder from SDL2
This commit is contained in:
Robert Beckebans
parent
0b66e57359
commit
5bef3a327f
4 changed files with 603 additions and 39 deletions
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@ -4,6 +4,8 @@
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Doom 3 BFG Edition GPL Source Code
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Doom 3 BFG Edition GPL Source Code
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Copyright (C) 1993-2012 id Software LLC, a ZeniMax Media company.
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Copyright (C) 1993-2012 id Software LLC, a ZeniMax Media company.
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Copyright (C) 2013 Robert Beckebans
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Copyright (C) 2013 Robert Beckebans
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Copyright (C) 1997-2012 Sam Lantinga <slouken@libsdl.org> (MS ADPCM decoder)
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Copyright (c) 2011 Chris Robinson <chris.kcat@gmail.com> (OpenAL helpers)
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This file is part of the Doom 3 BFG Edition GPL Source Code ("Doom 3 BFG Edition Source Code").
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This file is part of the Doom 3 BFG Edition GPL Source Code ("Doom 3 BFG Edition Source Code").
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@ -260,28 +262,8 @@ void idSoundSample_OpenAL::LoadResource()
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}
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}
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}
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}
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// build OpenAL buffer
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// upload PCM data to OpenAL
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alGetError();
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CreateOpenALBuffer();
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alGenBuffers( 1, &openalBuffer );
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if( alGetError() != AL_NO_ERROR )
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{
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common->Error( "idSoundSample_OpenAL::MakeDefault: error generating OpenAL hardware buffer" );
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}
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if( alIsBuffer( openalBuffer ) )
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{
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alGetError();
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// RB: TODO decode idWaveFile::FORMAT_ADPCM to idWaveFile::FORMAT_PCM
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// and build one big OpenAL buffer using the alBufferSubData extension
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alBufferData( openalBuffer, GetOpenALBufferFormat(), buffers[0].buffer, buffers[0].bufferSize, format.basic.samplesPerSec );
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if( alGetError() != AL_NO_ERROR )
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{
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common->Error( "idSoundSample_OpenAL::MakeDefault: error loading data into OpenAL hardware buffer" );
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}
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}
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return;
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return;
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}
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}
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@ -295,6 +277,88 @@ void idSoundSample_OpenAL::LoadResource()
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return;
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return;
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}
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}
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void idSoundSample_OpenAL::CreateOpenALBuffer()
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{
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// build OpenAL buffer
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alGetError();
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alGenBuffers( 1, &openalBuffer );
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if( alGetError() != AL_NO_ERROR )
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{
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common->Error( "idSoundSample_OpenAL::CreateOpenALBuffer: error generating OpenAL hardware buffer" );
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}
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if( alIsBuffer( openalBuffer ) )
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{
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alGetError();
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// RB: TODO decode idWaveFile::FORMAT_ADPCM to idWaveFile::FORMAT_PCM
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// and build one big OpenAL buffer using the alBufferSubData extension
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void* buffer = NULL;
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uint32 bufferSize = 0;
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if( format.basic.formatTag == idWaveFile::FORMAT_ADPCM )
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{
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buffer = buffers[0].buffer;
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bufferSize = buffers[0].bufferSize;
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if( MS_ADPCM_decode( ( uint8** ) &buffer, &bufferSize ) < 0 )
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{
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common->Error( "idSoundSample_OpenAL::CreateOpenALBuffer: could not decode ADPCM '%s' to 16 bit format", GetName() );
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}
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buffers[0].buffer = buffer;
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buffers[0].bufferSize = bufferSize;
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totalBufferSize = bufferSize;
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}
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else if( format.basic.formatTag == idWaveFile::FORMAT_XMA2 )
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{
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common->Error( "idSoundSample_OpenAL::CreateOpenALBuffer: could not decode XMA2 '%s' to 16 bit format", GetName() );
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}
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else if( format.basic.formatTag == idWaveFile::FORMAT_EXTENSIBLE )
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{
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common->Error( "idSoundSample_OpenAL::CreateOpenALBuffer: could not decode extensible WAV format '%s' to 16 bit format", GetName() );
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}
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else
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{
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// TODO concatenate buffers
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assert( buffers.Num() == 1 );
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buffer = buffers[0].buffer;
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bufferSize = buffers[0].bufferSize;
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}
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#if 0
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if( alIsExtensionPresent( "AL_SOFT_buffer_samples" ) )
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{
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ALenum type = AL_SHORT_SOFT;
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if( format.basic.bitsPerSample != 16 )
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{
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//common->Error( "idSoundSample_OpenAL::LoadResource: '%s' not a 16 bit format", GetName() );
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}
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ALenum channels = NumChannels() == 1 ? AL_MONO_SOFT : AL_STEREO_SOFT;
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ALenum alFormat = GetOpenALSoftFormat( channels, type );
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alBufferSamplesSOFT( openalBuffer, format.basic.samplesPerSec, alFormat, BytesToFrames( bufferSize, channels, type ), channels, type, buffer );
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}
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else
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#endif
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{
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alBufferData( openalBuffer, GetOpenALBufferFormat(), buffer, bufferSize, format.basic.samplesPerSec );
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}
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if( alGetError() != AL_NO_ERROR )
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{
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common->Error( "idSoundSample_OpenAL::CreateOpenALBuffer: error loading data into OpenAL hardware buffer" );
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}
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}
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}
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/*
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/*
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========================
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========================
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idSoundSample_OpenAL::LoadWav
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idSoundSample_OpenAL::LoadWav
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@ -618,18 +682,318 @@ float idSoundSample_OpenAL::GetAmplitude( int timeMS ) const
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}
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}
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const char* idSoundSample_OpenAL::OpenALSoftChannelsName( ALenum chans ) const
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{
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switch( chans )
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{
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case AL_MONO_SOFT:
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return "Mono";
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case AL_STEREO_SOFT:
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return "Stereo";
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case AL_REAR_SOFT:
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return "Rear";
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case AL_QUAD_SOFT:
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return "Quadraphonic";
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case AL_5POINT1_SOFT:
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return "5.1 Surround";
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case AL_6POINT1_SOFT:
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return "6.1 Surround";
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case AL_7POINT1_SOFT:
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return "7.1 Surround";
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}
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return "Unknown Channels";
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}
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const char* idSoundSample_OpenAL::OpenALSoftTypeName( ALenum type ) const
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{
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switch( type )
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{
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case AL_BYTE_SOFT:
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return "S8";
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case AL_UNSIGNED_BYTE_SOFT:
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return "U8";
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case AL_SHORT_SOFT:
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return "S16";
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case AL_UNSIGNED_SHORT_SOFT:
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return "U16";
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case AL_INT_SOFT:
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return "S32";
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case AL_UNSIGNED_INT_SOFT:
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return "U32";
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case AL_FLOAT_SOFT:
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return "Float32";
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case AL_DOUBLE_SOFT:
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return "Float64";
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}
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return "Unknown Type";
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}
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ALsizei idSoundSample_OpenAL::FramesToBytes( ALsizei size, ALenum channels, ALenum type ) const
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{
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switch( channels )
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{
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case AL_MONO_SOFT:
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size *= 1;
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break;
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case AL_STEREO_SOFT:
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size *= 2;
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break;
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case AL_REAR_SOFT:
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size *= 2;
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break;
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case AL_QUAD_SOFT:
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size *= 4;
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break;
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case AL_5POINT1_SOFT:
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size *= 6;
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break;
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case AL_6POINT1_SOFT:
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size *= 7;
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break;
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case AL_7POINT1_SOFT:
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size *= 8;
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break;
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}
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switch( type )
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{
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case AL_BYTE_SOFT:
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size *= sizeof( ALbyte );
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break;
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case AL_UNSIGNED_BYTE_SOFT:
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size *= sizeof( ALubyte );
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break;
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case AL_SHORT_SOFT:
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size *= sizeof( ALshort );
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break;
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case AL_UNSIGNED_SHORT_SOFT:
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size *= sizeof( ALushort );
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break;
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case AL_INT_SOFT:
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size *= sizeof( ALint );
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break;
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case AL_UNSIGNED_INT_SOFT:
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size *= sizeof( ALuint );
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break;
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case AL_FLOAT_SOFT:
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size *= sizeof( ALfloat );
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break;
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case AL_DOUBLE_SOFT:
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size *= sizeof( ALdouble );
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break;
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}
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return size;
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}
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ALsizei idSoundSample_OpenAL::BytesToFrames( ALsizei size, ALenum channels, ALenum type ) const
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{
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return size / FramesToBytes( 1, channels, type );
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}
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ALenum idSoundSample_OpenAL::GetOpenALSoftFormat( ALenum channels, ALenum type ) const
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{
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ALenum format = AL_NONE;
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/* If using AL_SOFT_buffer_samples, try looking through its formats */
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if( alIsExtensionPresent( "AL_SOFT_buffer_samples" ) )
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{
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/* AL_SOFT_buffer_samples is more lenient with matching formats. The
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* specified sample type does not need to match the returned format,
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* but it is nice to try to get something close. */
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if( type == AL_UNSIGNED_BYTE_SOFT || type == AL_BYTE_SOFT )
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{
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if( channels == AL_MONO_SOFT ) format = AL_MONO8_SOFT;
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else if( channels == AL_STEREO_SOFT ) format = AL_STEREO8_SOFT;
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else if( channels == AL_QUAD_SOFT ) format = AL_QUAD8_SOFT;
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else if( channels == AL_5POINT1_SOFT ) format = AL_5POINT1_8_SOFT;
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else if( channels == AL_6POINT1_SOFT ) format = AL_6POINT1_8_SOFT;
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else if( channels == AL_7POINT1_SOFT ) format = AL_7POINT1_8_SOFT;
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}
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else if( type == AL_UNSIGNED_SHORT_SOFT || type == AL_SHORT_SOFT )
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{
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if( channels == AL_MONO_SOFT ) format = AL_MONO16_SOFT;
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else if( channels == AL_STEREO_SOFT ) format = AL_STEREO16_SOFT;
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else if( channels == AL_QUAD_SOFT ) format = AL_QUAD16_SOFT;
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else if( channels == AL_5POINT1_SOFT ) format = AL_5POINT1_16_SOFT;
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else if( channels == AL_6POINT1_SOFT ) format = AL_6POINT1_16_SOFT;
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else if( channels == AL_7POINT1_SOFT ) format = AL_7POINT1_16_SOFT;
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}
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else if( type == AL_UNSIGNED_BYTE3_SOFT || type == AL_BYTE3_SOFT ||
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type == AL_UNSIGNED_INT_SOFT || type == AL_INT_SOFT ||
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type == AL_FLOAT_SOFT || type == AL_DOUBLE_SOFT )
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{
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if( channels == AL_MONO_SOFT ) format = AL_MONO32F_SOFT;
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else if( channels == AL_STEREO_SOFT ) format = AL_STEREO32F_SOFT;
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else if( channels == AL_QUAD_SOFT ) format = AL_QUAD32F_SOFT;
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else if( channels == AL_5POINT1_SOFT ) format = AL_5POINT1_32F_SOFT;
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else if( channels == AL_6POINT1_SOFT ) format = AL_6POINT1_32F_SOFT;
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else if( channels == AL_7POINT1_SOFT ) format = AL_7POINT1_32F_SOFT;
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}
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if( format != AL_NONE && !alIsBufferFormatSupportedSOFT( format ) )
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format = AL_NONE;
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/* A matching format was not found or supported. Try 32-bit float. */
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if( format == AL_NONE )
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{
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if( channels == AL_MONO_SOFT ) format = AL_MONO32F_SOFT;
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else if( channels == AL_STEREO_SOFT ) format = AL_STEREO32F_SOFT;
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else if( channels == AL_QUAD_SOFT ) format = AL_QUAD32F_SOFT;
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else if( channels == AL_5POINT1_SOFT ) format = AL_5POINT1_32F_SOFT;
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else if( channels == AL_6POINT1_SOFT ) format = AL_6POINT1_32F_SOFT;
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else if( channels == AL_7POINT1_SOFT ) format = AL_7POINT1_32F_SOFT;
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if( format != AL_NONE && !alIsBufferFormatSupportedSOFT( format ) )
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format = AL_NONE;
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}
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/* 32-bit float not supported. Try 16-bit int. */
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if( format == AL_NONE )
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{
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if( channels == AL_MONO_SOFT ) format = AL_MONO16_SOFT;
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else if( channels == AL_STEREO_SOFT ) format = AL_STEREO16_SOFT;
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else if( channels == AL_QUAD_SOFT ) format = AL_QUAD16_SOFT;
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else if( channels == AL_5POINT1_SOFT ) format = AL_5POINT1_16_SOFT;
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else if( channels == AL_6POINT1_SOFT ) format = AL_6POINT1_16_SOFT;
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else if( channels == AL_7POINT1_SOFT ) format = AL_7POINT1_16_SOFT;
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if( format != AL_NONE && !alIsBufferFormatSupportedSOFT( format ) )
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format = AL_NONE;
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}
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/* 16-bit int not supported. Try 8-bit int. */
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if( format == AL_NONE )
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{
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if( channels == AL_MONO_SOFT ) format = AL_MONO8_SOFT;
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else if( channels == AL_STEREO_SOFT ) format = AL_STEREO8_SOFT;
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else if( channels == AL_QUAD_SOFT ) format = AL_QUAD8_SOFT;
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else if( channels == AL_5POINT1_SOFT ) format = AL_5POINT1_8_SOFT;
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else if( channels == AL_6POINT1_SOFT ) format = AL_6POINT1_8_SOFT;
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else if( channels == AL_7POINT1_SOFT ) format = AL_7POINT1_8_SOFT;
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if( format != AL_NONE && !alIsBufferFormatSupportedSOFT( format ) )
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format = AL_NONE;
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||||||
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}
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return format;
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}
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/* We use the AL_EXT_MCFORMATS extension to provide output of Quad, 5.1,
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* and 7.1 channel configs, AL_EXT_FLOAT32 for 32-bit float samples, and
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* AL_EXT_DOUBLE for 64-bit float samples. */
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if( type == AL_UNSIGNED_BYTE_SOFT )
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{
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if( channels == AL_MONO_SOFT )
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format = AL_FORMAT_MONO8;
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else if( channels == AL_STEREO_SOFT )
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format = AL_FORMAT_STEREO8;
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else if( alIsExtensionPresent( "AL_EXT_MCFORMATS" ) )
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{
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if( channels == AL_QUAD_SOFT )
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format = alGetEnumValue( "AL_FORMAT_QUAD8" );
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else if( channels == AL_5POINT1_SOFT )
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format = alGetEnumValue( "AL_FORMAT_51CHN8" );
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else if( channels == AL_6POINT1_SOFT )
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format = alGetEnumValue( "AL_FORMAT_61CHN8" );
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else if( channels == AL_7POINT1_SOFT )
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format = alGetEnumValue( "AL_FORMAT_71CHN8" );
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}
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}
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else if( type == AL_SHORT_SOFT )
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{
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if( channels == AL_MONO_SOFT )
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format = AL_FORMAT_MONO16;
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else if( channels == AL_STEREO_SOFT )
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format = AL_FORMAT_STEREO16;
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||||||
|
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 idSoundSample_OpenAL::GetOpenALBufferFormat() const
|
||||||
{
|
{
|
||||||
ALenum alFormat;
|
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 )
|
if( format.basic.formatTag == idWaveFile::FORMAT_PCM )
|
||||||
{
|
{
|
||||||
alFormat = NumChannels() == 1 ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16;
|
alFormat = NumChannels() == 1 ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16;
|
||||||
}
|
}
|
||||||
else if( format.basic.formatTag == idWaveFile::FORMAT_ADPCM )
|
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_MONO16 : AL_FORMAT_STEREO16;
|
||||||
//alFormat = NumChannels() == 1 ? AL_FORMAT_IMA_ADPCM_MONO16_EXT : AL_FORMAT_IMA_ADPCM_STEREO16_EXT;
|
//alFormat = NumChannels() == 1 ? AL_FORMAT_MONO_IMA4 : AL_FORMAT_STEREO_IMA4;
|
||||||
}
|
}
|
||||||
else if( format.basic.formatTag == idWaveFile::FORMAT_XMA2 )
|
else if( format.basic.formatTag == idWaveFile::FORMAT_XMA2 )
|
||||||
{
|
{
|
||||||
|
|
@ -639,7 +1003,177 @@ ALenum idSoundSample_OpenAL::GetOpenALBufferFormat() const
|
||||||
{
|
{
|
||||||
alFormat = NumChannels() == 1 ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16;
|
alFormat = NumChannels() == 1 ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16;
|
||||||
}
|
}
|
||||||
|
}
|
||||||
|
|
||||||
return alFormat;
|
return alFormat;
|
||||||
}
|
}
|
||||||
|
|
||||||
|
int32 idSoundSample_OpenAL::MS_ADPCM_nibble( MS_ADPCM_decodeState_t* state, int8 nybble, int16* coeff )
|
||||||
|
{
|
||||||
|
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 * coeff[0] ) +
|
||||||
|
( state->iSamp2 * coeff[1] ) ) / 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 state[2];
|
||||||
|
uint8* freeable, *encoded, *decoded;
|
||||||
|
int32 encoded_len, samplesleft;
|
||||||
|
int8 nybble;
|
||||||
|
bool stereo;
|
||||||
|
int16* coeff[2];
|
||||||
|
int32 new_sample;
|
||||||
|
|
||||||
|
// Allocate the proper sized output buffer
|
||||||
|
encoded_len = *audio_len;
|
||||||
|
encoded = *audio_buf;
|
||||||
|
freeable = *audio_buf;
|
||||||
|
|
||||||
|
/*
|
||||||
|
*audio_len = ( encoded_len / MS_ADPCM_state.wavefmt.blockalign ) *
|
||||||
|
MS_ADPCM_state.wSamplesPerBlock *
|
||||||
|
MS_ADPCM_state.wavefmt.channels * sizeof( int16 );
|
||||||
|
*/
|
||||||
|
|
||||||
|
*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;
|
||||||
|
|
||||||
|
// Get ready... Go!
|
||||||
|
stereo = ( format.basic.numChannels == 2 );
|
||||||
|
//state[0] = &MS_ADPCM_state.state[0];
|
||||||
|
//state[1] = &MS_ADPCM_state.state[stereo];
|
||||||
|
|
||||||
|
while( encoded_len >= format.basic.blockSize )
|
||||||
|
{
|
||||||
|
// Grab the initial information for this block
|
||||||
|
state[0].hPredictor = *encoded++;
|
||||||
|
if( stereo )
|
||||||
|
{
|
||||||
|
state[1].hPredictor = *encoded++;
|
||||||
|
}
|
||||||
|
|
||||||
|
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 );
|
||||||
|
}
|
||||||
|
|
||||||
|
assert( state[0].hPredictor < format.extra.adpcm.numCoef );
|
||||||
|
assert( state[1].hPredictor < format.extra.adpcm.numCoef );
|
||||||
|
|
||||||
|
coeff[0] = &format.extra.adpcm.aCoef[state[0].hPredictor].coef1;
|
||||||
|
coeff[1] = &format.extra.adpcm.aCoef[state[1].hPredictor].coef2;
|
||||||
|
|
||||||
|
// Store the two initial samples we start with
|
||||||
|
decoded[0] = state[0].iSamp2 & 0xFF;
|
||||||
|
decoded[1] = state[0].iSamp2 >> 8;
|
||||||
|
decoded += 2;
|
||||||
|
if( stereo )
|
||||||
|
{
|
||||||
|
decoded[0] = state[1].iSamp2 & 0xFF;
|
||||||
|
decoded[1] = state[1].iSamp2 >> 8;
|
||||||
|
decoded += 2;
|
||||||
|
}
|
||||||
|
decoded[0] = state[0].iSamp1 & 0xFF;
|
||||||
|
decoded[1] = state[0].iSamp1 >> 8;
|
||||||
|
decoded += 2;
|
||||||
|
if( stereo )
|
||||||
|
{
|
||||||
|
decoded[0] = state[1].iSamp1 & 0xFF;
|
||||||
|
decoded[1] = state[1].iSamp1 >> 8;
|
||||||
|
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, coeff[0] );
|
||||||
|
decoded[0] = new_sample & 0xFF;
|
||||||
|
new_sample >>= 8;
|
||||||
|
decoded[1] = new_sample & 0xFF;
|
||||||
|
decoded += 2;
|
||||||
|
|
||||||
|
nybble = ( *encoded ) & 0x0F;
|
||||||
|
new_sample = MS_ADPCM_nibble( &state[1], nybble, coeff[1] );
|
||||||
|
decoded[0] = new_sample & 0xFF;
|
||||||
|
new_sample >>= 8;
|
||||||
|
decoded[1] = new_sample & 0xFF;
|
||||||
|
decoded += 2;
|
||||||
|
|
||||||
|
++encoded;
|
||||||
|
samplesleft -= 2;
|
||||||
|
}
|
||||||
|
|
||||||
|
encoded_len -= format.basic.blockSize;
|
||||||
|
}
|
||||||
|
|
||||||
|
Mem_Free( freeable );
|
||||||
|
|
||||||
|
return 0;
|
||||||
|
}
|
||||||
|
|
||||||
|
|
|
||||||
|
|
@ -130,8 +130,24 @@ public:
|
||||||
|
|
||||||
float GetAmplitude( int timeMS ) const;
|
float GetAmplitude( int timeMS ) const;
|
||||||
|
|
||||||
|
const char* OpenALSoftChannelsName( ALenum chans ) const;
|
||||||
|
|
||||||
|
const char* OpenALSoftTypeName( ALenum type ) const;
|
||||||
|
|
||||||
|
ALsizei FramesToBytes( ALsizei size, ALenum channels, ALenum type ) const;
|
||||||
|
ALsizei BytesToFrames( ALsizei size, ALenum channels, ALenum type ) const;
|
||||||
|
|
||||||
|
/* Retrieves a compatible buffer format given the channel configuration and
|
||||||
|
* sample type. If an alIsBufferFormatSupportedSOFT-compatible function is
|
||||||
|
* provided, it will be called to find the closest-matching format from
|
||||||
|
* AL_SOFT_buffer_samples. Returns AL_NONE (0) if no supported format can be
|
||||||
|
* found. */
|
||||||
|
ALenum GetOpenALSoftFormat( ALenum channels, ALenum type ) const;
|
||||||
|
|
||||||
ALenum GetOpenALBufferFormat() const;
|
ALenum GetOpenALBufferFormat() const;
|
||||||
|
|
||||||
|
void CreateOpenALBuffer();
|
||||||
|
|
||||||
protected:
|
protected:
|
||||||
friend class idSoundHardware_OpenAL;
|
friend class idSoundHardware_OpenAL;
|
||||||
friend class idSoundVoice_OpenAL;
|
friend class idSoundVoice_OpenAL;
|
||||||
|
|
@ -144,6 +160,17 @@ protected:
|
||||||
bool LoadGeneratedSample( const idStr& name );
|
bool LoadGeneratedSample( const idStr& name );
|
||||||
void WriteGeneratedSample( idFile* fileOut );
|
void WriteGeneratedSample( idFile* fileOut );
|
||||||
|
|
||||||
|
struct MS_ADPCM_decodeState_t
|
||||||
|
{
|
||||||
|
uint8 hPredictor;
|
||||||
|
uint16 iDelta;
|
||||||
|
int16 iSamp1;
|
||||||
|
int16 iSamp2;
|
||||||
|
};
|
||||||
|
|
||||||
|
int32 MS_ADPCM_nibble( MS_ADPCM_decodeState_t* state, int8 nybble, int16* coeff );
|
||||||
|
int MS_ADPCM_decode( uint8** audio_buf, uint32* audio_len );
|
||||||
|
|
||||||
struct sampleBuffer_t
|
struct sampleBuffer_t
|
||||||
{
|
{
|
||||||
void* buffer;
|
void* buffer;
|
||||||
|
|
|
||||||
|
|
@ -31,7 +31,7 @@ If you have questions concerning this license or the applicable additional terms
|
||||||
#include "../snd_local.h"
|
#include "../snd_local.h"
|
||||||
|
|
||||||
idCVar s_skipHardwareSets( "s_skipHardwareSets", "0", CVAR_BOOL, "Do all calculation, but skip XA2 calls" );
|
idCVar s_skipHardwareSets( "s_skipHardwareSets", "0", CVAR_BOOL, "Do all calculation, but skip XA2 calls" );
|
||||||
idCVar s_debugHardware( "s_debugHardware", "1", CVAR_BOOL, "Print a message any time a hardware voice changes" );
|
idCVar s_debugHardware( "s_debugHardware", "0", CVAR_BOOL, "Print a message any time a hardware voice changes" );
|
||||||
|
|
||||||
// The whole system runs at this sample rate
|
// The whole system runs at this sample rate
|
||||||
static int SYSTEM_SAMPLE_RATE = 44100;
|
static int SYSTEM_SAMPLE_RATE = 44100;
|
||||||
|
|
@ -170,10 +170,11 @@ void idSoundVoice_OpenAL::Create( const idSoundSample* leadinSample_, const idSo
|
||||||
*/
|
*/
|
||||||
}
|
}
|
||||||
else
|
else
|
||||||
|
{
|
||||||
|
//if( triggered )
|
||||||
|
|
||||||
// handle streaming sounds (decode on the fly) both single shot AND looping
|
// handle streaming sounds (decode on the fly) both single shot AND looping
|
||||||
//if( triggered )
|
|
||||||
{
|
|
||||||
alSourcei( openalSource, AL_BUFFER, 0 );
|
alSourcei( openalSource, AL_BUFFER, 0 );
|
||||||
alDeleteBuffers( 3, &lastopenalStreamingBuffer[0] );
|
alDeleteBuffers( 3, &lastopenalStreamingBuffer[0] );
|
||||||
lastopenalStreamingBuffer[0] = openalStreamingBuffer[0];
|
lastopenalStreamingBuffer[0] = openalStreamingBuffer[0];
|
||||||
|
|
@ -213,7 +214,7 @@ void idSoundVoice_OpenAL::Create( const idSoundSample* leadinSample_, const idSo
|
||||||
alSource3f( openalSource, AL_POSITION, 0.0f, 0.0f, 0.0f );
|
alSource3f( openalSource, AL_POSITION, 0.0f, 0.0f, 0.0f );
|
||||||
|
|
||||||
// RB: FIXME 0.0f ?
|
// RB: FIXME 0.0f ?
|
||||||
alSourcef( openalSource, AL_GAIN, 0.0f );
|
alSourcef( openalSource, AL_GAIN, 1.0f );
|
||||||
|
|
||||||
//OnBufferStart( leadinSample, 0 );
|
//OnBufferStart( leadinSample, 0 );
|
||||||
}
|
}
|
||||||
|
|
|
||||||
|
|
@ -95,6 +95,8 @@ typedef enum
|
||||||
|
|
||||||
#if defined(USE_OPENAL)
|
#if defined(USE_OPENAL)
|
||||||
|
|
||||||
|
#define AL_ALEXT_PROTOTYPES
|
||||||
|
|
||||||
#include <AL/al.h>
|
#include <AL/al.h>
|
||||||
#include <AL/alc.h>
|
#include <AL/alc.h>
|
||||||
#include <AL/alext.h>
|
#include <AL/alext.h>
|
||||||
|
|
|
||||||
Loading…
Reference in a new issue