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Fixed problems in the ADPCM decoder. Sound is working.

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This commit is contained in:
Robert Beckebans 2013-01-05 19:00:22 +01:00
parent 27b83db50d
commit 699e0e39f5
2 changed files with 65 additions and 52 deletions
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112
neo/sound/OpenAL/AL_SoundSample.cpp
View file

@ -1008,7 +1008,7 @@ ALenum idSoundSample_OpenAL::GetOpenALBufferFormat() const
return alFormat;
}
int32 idSoundSample_OpenAL::MS_ADPCM_nibble( MS_ADPCM_decodeState_t* state, int8 nybble, int16* coeff )
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 ) );
@ -1017,10 +1017,11 @@ int32 idSoundSample_OpenAL::MS_ADPCM_nibble( MS_ADPCM_decodeState_t* state, int8
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;
new_sample = ( ( state->iSamp1 * state->coef1 ) +
( state->iSamp2 * state->coef2 ) ) / 256;
if( nybble & 0x08 )
{
@ -1055,12 +1056,13 @@ int32 idSoundSample_OpenAL::MS_ADPCM_nibble( MS_ADPCM_decodeState_t* state, int8
int idSoundSample_OpenAL::MS_ADPCM_decode( uint8** audio_buf, uint32* audio_len )
{
static MS_ADPCM_decodeState_t state[2];
static MS_ADPCM_decodeState_t states[2];
MS_ADPCM_decodeState_t* state[2];
uint8* freeable, *encoded, *decoded;
int32 encoded_len, samplesleft;
int8 nybble;
bool stereo;
int16* coeff[2];
int8 stereo;
int32 new_sample;
// Allocate the proper sized output buffer
@ -1068,12 +1070,6 @@ int idSoundSample_OpenAL::MS_ADPCM_decode( uint8** audio_buf, uint32* 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 );
@ -1084,65 +1080,79 @@ int idSoundSample_OpenAL::MS_ADPCM_decode( uint8** audio_buf, uint32* audio_len
}
decoded = *audio_buf;
assert( format.basic.numChannels == 1 || format.basic.numChannels == 2 );
// Get ready... Go!
stereo = ( format.basic.numChannels == 2 );
//state[0] = &MS_ADPCM_state.state[0];
//state[1] = &MS_ADPCM_state.state[stereo];
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++;
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++;
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] );
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] );
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[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;
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;
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;
decoded[0] = state[1]->iSamp1 & 0xFF;
decoded[1] = ( state[1]->iSamp1 >> 8 ) & 0xFF;
decoded += 2;
}
@ -1152,17 +1162,17 @@ int idSoundSample_OpenAL::MS_ADPCM_decode( uint8** audio_buf, uint32* audio_len
while( samplesleft > 0 )
{
nybble = ( *encoded ) >> 4;
new_sample = MS_ADPCM_nibble( &state[0], nybble, coeff[0] );
new_sample = MS_ADPCM_nibble( state[0], nybble );
decoded[0] = new_sample & 0xFF;
new_sample >>= 8;
decoded[1] = new_sample & 0xFF;
decoded[1] = ( new_sample >> 8 ) & 0xFF;
decoded += 2;
nybble = ( *encoded ) & 0x0F;
new_sample = MS_ADPCM_nibble( &state[1], nybble, coeff[1] );
new_sample = MS_ADPCM_nibble( state[1], nybble );
decoded[0] = new_sample & 0xFF;
new_sample >>= 8;
decoded[1] = new_sample & 0xFF;
decoded[1] = ( new_sample >> 8 ) & 0xFF;
decoded += 2;
++encoded;

5
neo/sound/OpenAL/AL_SoundSample.h
View file

@ -163,12 +163,15 @@ protected:
struct MS_ADPCM_decodeState_t
{
uint8 hPredictor;
int16 coef1;
int16 coef2;
uint16 iDelta;
int16 iSamp1;
int16 iSamp2;
};
int32 MS_ADPCM_nibble( MS_ADPCM_decodeState_t* state, int8 nybble, int16* coeff );
int32 MS_ADPCM_nibble( MS_ADPCM_decodeState_t* state, int8 nybble );
int MS_ADPCM_decode( uint8** audio_buf, uint32* audio_len );
struct sampleBuffer_t