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mirror of https://github.com/AxioDL/metaforce.git synced 2025-12-10 00:27:43 +00:00

THP audio fix

This commit is contained in:
Jack Andersen
2016-03-09 10:03:35 -10:00
parent 5f7c6769e3
commit f53bdcc5bd
5 changed files with 36 additions and 36 deletions

View File

@@ -1,26 +1,26 @@
#include "dsp.h"
static const int NibbleToInt[16] = {0,1,2,3,4,5,6,7,-8,-7,-6,-5,-4,-3,-2,-1};
static const int32_t NibbleToInt[16] = {0,1,2,3,4,5,6,7,-8,-7,-6,-5,-4,-3,-2,-1};
void DSPDecompressFrame(int16_t* out, const uint8_t* in,
const int16_t coefs[8][2], int16_t* prev1, int16_t* prev2,
unsigned lastSample)
{
uint8_t cIdx = (in[0]>>4) & 0xf;
int16_t factor1 = coefs[cIdx][0];
int16_t factor2 = coefs[cIdx][1];
int32_t factor1 = coefs[cIdx][0];
int32_t factor2 = coefs[cIdx][1];
uint8_t exp = in[0] & 0xf;
for (int s=0 ; s<14 && s<lastSample ; ++s)
{
int sampleData = (s&1)?
int32_t sampleData = (s&1)?
NibbleToInt[(in[s/2+1])&0xf]:
NibbleToInt[(in[s/2+1]>>4)&0xf];
sampleData <<= exp;
sampleData <<= 11;
sampleData += 1024;
sampleData +=
factor1 * *prev1 +
factor2 * *prev2;
factor1 * ((int32_t)(*prev1)) +
factor2 * ((int32_t)(*prev2));
sampleData >>= 11;
sampleData = DSPSampClamp(sampleData);
out[s] = sampleData;
@@ -33,10 +33,10 @@ void DSPDecompressFrameStereoStride(int16_t* out, const uint8_t* in,
const int16_t coefs[8][2], int16_t* prev1, int16_t* prev2,
unsigned lastSample)
{
uint8_t cIdx = (in[0]>>4) & 0xf;
int16_t factor1 = coefs[cIdx][0];
int16_t factor2 = coefs[cIdx][1];
uint8_t exp = in[0] & 0xf;
uint32_t cIdx = (in[0]>>4) & 0xf;
int32_t factor1 = coefs[cIdx][0];
int32_t factor2 = coefs[cIdx][1];
uint32_t exp = in[0] & 0xf;
for (int s=0 ; s<14 && s<lastSample ; ++s)
{
int sampleData = (s&1)?
@@ -46,8 +46,8 @@ void DSPDecompressFrameStereoStride(int16_t* out, const uint8_t* in,
sampleData <<= 11;
sampleData += 1024;
sampleData +=
factor1 * *prev1 +
factor2 * *prev2;
factor1 * ((int32_t)(*prev1)) +
factor2 * ((int32_t)(*prev2));
sampleData >>= 11;
sampleData = DSPSampClamp(sampleData);
out[s*2] = sampleData;
@@ -61,8 +61,8 @@ void DSPDecompressFrameStereoDupe(int16_t* out, const uint8_t* in,
unsigned lastSample)
{
uint8_t cIdx = (in[0]>>4) & 0xf;
int16_t factor1 = coefs[cIdx][0];
int16_t factor2 = coefs[cIdx][1];
int32_t factor1 = coefs[cIdx][0];
int32_t factor2 = coefs[cIdx][1];
uint8_t exp = in[0] & 0xf;
for (int s=0 ; s<14 && s<lastSample ; ++s)
{
@@ -73,8 +73,8 @@ void DSPDecompressFrameStereoDupe(int16_t* out, const uint8_t* in,
sampleData <<= 11;
sampleData += 1024;
sampleData +=
factor1 * *prev1 +
factor2 * *prev2;
factor1 * ((int32_t)(*prev1)) +
factor2 * ((int32_t)(*prev2));
sampleData >>= 11;
sampleData = DSPSampClamp(sampleData);
out[s*2] = sampleData;

View File

@@ -7,10 +7,10 @@ extern "C" {
#include <stdint.h>
static inline short DSPSampClamp(int val)
static inline short DSPSampClamp(int32_t val)
{
if (val < -32768) val = -32768;
if (val > 32767) val = 32767;
else if (val > 32767) val = 32767;
return val;
}

View File

@@ -310,16 +310,15 @@ u32 CMoviePlayer::THPAudioDecode(s16* buffer, const u8* audioFrame, bool stereo)
{
unsigned samples = header.numSamples;
s16* bufferCur = buffer+i;
const u8* audioFrameCur = audioFrame;
int16_t prev1 = header.channelPrevs[i][0];
int16_t prev2 = header.channelPrevs[i][1];
for (int f=0 ; f<header.channelSize/8 ; ++f)
for (int f=0 ; f<(header.numSamples + 13) / 14 ; ++f)
{
DSPDecompressFrameStereoStride(bufferCur, audioFrameCur,
DSPDecompressFrameStereoStride(bufferCur, audioFrame,
header.channelCoefs[i], &prev1, &prev2, samples);
samples -= 14;
bufferCur += 28;
audioFrameCur += 8;
audioFrame += 8;
}
}
}
@@ -327,16 +326,15 @@ u32 CMoviePlayer::THPAudioDecode(s16* buffer, const u8* audioFrame, bool stereo)
{
unsigned samples = header.numSamples;
s16* bufferCur = buffer;
const u8* audioFrameCur = audioFrame;
int16_t prev1 = header.channelPrevs[0][0];
int16_t prev2 = header.channelPrevs[0][1];
for (int f=0 ; f<header.channelSize/8 ; ++f)
for (int f=0 ; f<(header.numSamples + 13) / 14 ; ++f)
{
DSPDecompressFrameStereoDupe(bufferCur, audioFrameCur,
DSPDecompressFrameStereoDupe(bufferCur, audioFrame,
header.channelCoefs[0], &prev1, &prev2, samples);
samples -= 14;
bufferCur += 28;
audioFrameCur += 8;
audioFrame += 8;
}
}
@@ -524,25 +522,22 @@ void CMoviePlayer::MixAudio(s16* out, const s16* in, u32 samples)
{
if (in)
{
for (u32 i=0 ; i<thisSamples ; ++i)
for (u32 i=0 ; i<thisSamples ; ++i, out += 2, in += 2)
{
out[0] = DSPSampClamp(in[0] +
s16(s32(tex->audioBuf[(i+tex->playedSamples)*2]) * 0x50F4 / 0x8000));
out[1] = DSPSampClamp(in[1] +
s16(s32(tex->audioBuf[(i+tex->playedSamples)*2+1]) * 0x50F4 / 0x8000));
out += 2;
in += 2;
}
}
else
{
for (u32 i=0 ; i<thisSamples ; ++i)
for (u32 i=0 ; i<thisSamples ; ++i, out += 2)
{
out[0] = DSPSampClamp(
s16(s32(tex->audioBuf[(i+tex->playedSamples)*2]) * 0x50F4 / 0x8000));
out[1] = DSPSampClamp(
s16(s32(tex->audioBuf[(i+tex->playedSamples)*2+1]) * 0x50F4 / 0x8000));
out += 2;
}
}
tex->playedSamples += thisSamples;
@@ -554,6 +549,7 @@ void CMoviePlayer::MixAudio(s16* out, const s16* in, u32 samples)
memcpy(out, in, samples * 4);
else
memset(out, 0, samples * 4);
//fprintf(stderr, "dropped %d samples\n", samples);
return;
}
}
@@ -810,6 +806,7 @@ void CMoviePlayer::DecodeFromRead(const void* data)
break;
}
case THPComponents::Type::Audio:
memset(tex.audioBuf.get(), 0, x28_thpHead.maxAudioSamples * 4);
tex.audioSamples = THPAudioDecode(tex.audioBuf.get(), (u8*)inptr, x74_audioInfo.numChannels == 2);
tex.playedSamples = 0;
inptr += frameHeader.audioSize;