Implementing our own WAV decoder to replace SndFile

This commit is contained in:
Sergeanur 2021-01-06 15:46:59 +02:00
parent a6409fb445
commit 493f6cb578
2 changed files with 330 additions and 28 deletions

View File

@ -8,10 +8,14 @@
#include <opusfile.h>
#else
#ifdef _WIN32
#ifdef AUDIO_OAL_USE_SNDFILE
#pragma comment( lib, "libsndfile-1.lib" )
#endif
#pragma comment( lib, "libmpg123-0.lib" )
#endif
#ifdef AUDIO_OAL_USE_SNDFILE
#include <sndfile.h>
#endif
#include <mpg123.h>
#endif
@ -78,6 +82,290 @@ public:
CSortStereoBuffer SortStereoBuffer;
#ifndef AUDIO_OPUS
class CImaADPCMDecoder
{
const uint16 StepTable[89] = {
7, 8, 9, 10, 11, 12, 13, 14,
16, 17, 19, 21, 23, 25, 28, 31,
34, 37, 41, 45, 50, 55, 60, 66,
73, 80, 88, 97, 107, 118, 130, 143,
157, 173, 190, 209, 230, 253, 279, 307,
337, 371, 408, 449, 494, 544, 598, 658,
724, 796, 876, 963, 1060, 1166, 1282, 1411,
1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024,
3327, 3660, 4026, 4428, 4871, 5358, 5894, 6484,
7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794,
32767
};
int16 Sample, StepIndex;
public:
CImaADPCMDecoder()
{
Init(0, 0);
}
void Init(int16 _Sample, int16 _StepIndex)
{
Sample = _Sample;
StepIndex = _StepIndex;
}
void Decode(uint8 *inbuf, int16 *_outbuf, size_t size)
{
int16* outbuf = _outbuf;
for (size_t i = 0; i < size; i++)
{
*(outbuf++) = DecodeSample(inbuf[i] & 0xF);
*(outbuf++) = DecodeSample(inbuf[i] >> 4);
}
}
int16 DecodeSample(uint8 adpcm)
{
uint16 step = StepTable[StepIndex];
if (adpcm & 4)
StepIndex += ((adpcm & 3) + 1) * 2;
else
StepIndex--;
StepIndex = clamp(StepIndex, 0, 88);
int delta = step >> 3;
if (adpcm & 1) delta += step >> 2;
if (adpcm & 2) delta += step >> 1;
if (adpcm & 4) delta += step;
if (adpcm & 8) delta = -delta;
int newSample = Sample + delta;
Sample = clamp(newSample, -32768, 32767);
return Sample;
}
};
class CWavFile : public IDecoder
{
enum
{
WAVEFMT_PCM = 1,
WAVEFMT_IMA_ADPCM = 0x11,
WAVEFMT_XBOX_ADPCM = 0x69,
};
struct tDataHeader
{
uint32 ID;
uint32 Size;
};
struct tFormatHeader
{
uint16 AudioFormat;
uint16 NumChannels;
uint32 SampleRate;
uint32 ByteRate;
uint16 BlockAlign;
uint16 BitsPerSample;
uint16 extra[2]; // adpcm only
tFormatHeader() { memset(this, 0, sizeof(*this)); }
};
FILE* pFile;
bool bIsOpen;
tFormatHeader FormatHeader;
uint32 DataStartOffset;
uint32 SampleCount;
uint32 SamplesPerBlock;
// ADPCM things
uint8 *AdpcmBlock;
int16 **buffers;
CImaADPCMDecoder* decoders;
void Close()
{
if (pFile) {
fclose(pFile);
pFile = nil;
}
if (AdpcmBlock) delete AdpcmBlock;
if (buffers) delete buffers;
if (decoders) delete decoders;
}
public:
CWavFile(const char* path) : bIsOpen(false), DataStartOffset(0), SampleCount(0), SamplesPerBlock(0), AdpcmBlock(nil), buffers(nil), decoders(nil)
{
pFile = fopen(path, "rb");
if (!pFile) return;
#define CLOSE_ON_ERROR(op)\
if (op) { \
Close(); \
return; \
}
tDataHeader DataHeader;
CLOSE_ON_ERROR(fread(&DataHeader, sizeof(DataHeader), 1, pFile) == 0);
CLOSE_ON_ERROR(DataHeader.ID != 'FFIR');
int WAVE;
CLOSE_ON_ERROR(fread(&WAVE, 4, 1, pFile) == 0);
CLOSE_ON_ERROR(WAVE != 'EVAW')
CLOSE_ON_ERROR(fread(&DataHeader, sizeof(DataHeader), 1, pFile) == 0);
CLOSE_ON_ERROR(DataHeader.ID != ' tmf');
CLOSE_ON_ERROR(fread(&FormatHeader, Min(DataHeader.Size, sizeof(tFormatHeader)), 1, pFile) == 0);
CLOSE_ON_ERROR(DataHeader.Size > sizeof(tFormatHeader));
switch (FormatHeader.AudioFormat)
{
case WAVEFMT_XBOX_ADPCM:
FormatHeader.AudioFormat = WAVEFMT_IMA_ADPCM;
case WAVEFMT_IMA_ADPCM:
SamplesPerBlock = (FormatHeader.BlockAlign / FormatHeader.NumChannels - 4) * 2 + 1;
AdpcmBlock = new uint8[FormatHeader.BlockAlign];
buffers = new int16*[FormatHeader.NumChannels];
decoders = new CImaADPCMDecoder[FormatHeader.NumChannels];
break;
case WAVEFMT_PCM:
SamplesPerBlock = 1;
if (FormatHeader.BitsPerSample != 16)
{
debug("Unsupported PCM (%d bits), only signed 16-bit is supported (%s)\n", FormatHeader.BitsPerSample, path);
return;
}
break;
default:
debug("Unsupported wav format 0x%x (%s)\n", FormatHeader.AudioFormat, path);
return;
}
while (true) {
CLOSE_ON_ERROR(fread(&DataHeader, sizeof(DataHeader), 1, pFile) == 0);
if (DataHeader.ID == 'atad')
break;
fseek(pFile, DataHeader.Size, SEEK_CUR);
}
DataStartOffset = ftell(pFile);
SampleCount = DataHeader.Size / FormatHeader.BlockAlign * SamplesPerBlock;
bIsOpen = true;
#undef CLOSE_ON_ERROR
}
~CWavFile()
{
Close();
}
bool IsOpened()
{
return bIsOpen;
}
uint32 GetSampleSize()
{
return sizeof(uint16);
}
uint32 GetSampleCount()
{
return SampleCount;
}
uint32 GetSampleRate()
{
return FormatHeader.SampleRate;
}
uint32 GetChannels()
{
return FormatHeader.NumChannels;
}
void Seek(uint32 milliseconds)
{
if (!IsOpened()) return;
fseek(pFile, DataStartOffset + ms2samples(milliseconds) / SamplesPerBlock * FormatHeader.BlockAlign, SEEK_SET);
}
uint32 Tell()
{
if (!IsOpened()) return 0;
return samples2ms((ftell(pFile) - DataStartOffset) / FormatHeader.BlockAlign * SamplesPerBlock);
}
#define SAMPLES_IN_LINE (8)
uint32 Decode(void* buffer)
{
if (!IsOpened()) return 0;
if (FormatHeader.AudioFormat == WAVEFMT_PCM)
{
uint32 size = fread(buffer, 1, GetBufferSize(), pFile);
if (FormatHeader.NumChannels == 2)
SortStereoBuffer.SortStereo(buffer, size);
return size;
}
else if (FormatHeader.AudioFormat == WAVEFMT_IMA_ADPCM)
{
uint32 MaxSamples = GetBufferSamples() / FormatHeader.NumChannels;
uint32 CurSample = (ftell(pFile) - DataStartOffset) / FormatHeader.BlockAlign * SamplesPerBlock;
MaxSamples = Min(MaxSamples, SampleCount - CurSample);
MaxSamples = MaxSamples / SamplesPerBlock * SamplesPerBlock;
uint32 OutBufSizePerChannel = MaxSamples * GetSampleSize();
uint32 OutBufSize = OutBufSizePerChannel * FormatHeader.NumChannels;
int16** buffers = new int16*[FormatHeader.NumChannels];
CImaADPCMDecoder* decoders = new CImaADPCMDecoder[FormatHeader.NumChannels];
for (uint32 i = 0; i < FormatHeader.NumChannels; i++)
buffers[i] = (int16*)((int8*)buffer + OutBufSizePerChannel * i);
uint32 samplesRead = 0;
while (samplesRead < MaxSamples)
{
uint8* AdpcmBuf = AdpcmBlock;
if (fread(AdpcmBlock, 1, FormatHeader.BlockAlign, pFile) == 0)
return 0;
for (uint32 i = 0; i < FormatHeader.NumChannels; i++)
{
int16 Sample = *(int16*)AdpcmBuf;
AdpcmBuf += sizeof(int16);
int16 Step = *(int16*)AdpcmBuf;
AdpcmBuf += sizeof(int16);
decoders[i].Init(Sample, Step);
*(buffers[i]) = Sample;
buffers[i]++;
}
samplesRead++;
for (uint32 s = 1; s < SamplesPerBlock; s += SAMPLES_IN_LINE)
{
for (uint32 i = 0; i < FormatHeader.NumChannels; i++)
{
decoders[i].Decode(AdpcmBuf, buffers[i], SAMPLES_IN_LINE / 2);
AdpcmBuf += SAMPLES_IN_LINE / 2;
buffers[i] += SAMPLES_IN_LINE;
}
samplesRead += SAMPLES_IN_LINE;
}
}
return OutBufSize;
}
return 0;
}
};
#ifdef AUDIO_OAL_USE_SNDFILE
class CSndFile : public IDecoder
{
SNDFILE *m_pfSound;
@ -146,6 +434,7 @@ public:
return size;
}
};
#endif
#ifdef _WIN32
// fuzzy seek eliminates stutter when playing ADF but spams errors a lot (nothing breaks though)
@ -280,7 +569,7 @@ public:
static short quantize(double sample)
{
int a = int(sample + 0.5);
return short(clamp(int(sample + 0.5), -32768, 32767));
return short(clamp(a, -32768, 32767));
}
void Decode(void* _inbuf, int16* _outbuf, size_t size)
@ -336,6 +625,7 @@ class CVbFile : public IDecoder
size_t m_CurrentBlock;
uint8** ppTempBuffers;
int16** buffers;
void ReadBlock(int32 block = -1)
{
@ -349,10 +639,12 @@ class CVbFile : public IDecoder
}
public:
CVbFile(const char* path, uint32 nSampleRate = 32000, uint8 nChannels = 2) : m_nSampleRate(nSampleRate), m_nChannels(nChannels)
CVbFile(const char* path, uint32 nSampleRate = 32000, uint8 nChannels = 2) : m_nSampleRate(nSampleRate), m_nChannels(nChannels), decoders(nil), ppTempBuffers(nil), buffers(nil),
m_FileSize(0), m_nNumberOfBlocks(0), m_bBlockRead(false), m_LineInBlock(0), m_CurrentBlock(0)
{
pFile = fopen(path, "rb");
if (pFile) {
if (!pFile) return;
fseek(pFile, 0, SEEK_END);
m_FileSize = ftell(pFile);
fseek(pFile, 0, SEEK_SET);
@ -361,11 +653,11 @@ public:
m_CurrentBlock = 0;
m_LineInBlock = 0;
m_bBlockRead = false;
ppTempBuffers = new uint8 * [nChannels];
ppTempBuffers = new uint8*[nChannels];
buffers = new int16*[nChannels];
for (uint8 i = 0; i < nChannels; i++)
ppTempBuffers[i] = new uint8[VB_BLOCK_SIZE];
}
}
~CVbFile()
{
@ -376,6 +668,7 @@ public:
for (int i = 0; i < m_nChannels; i++)
delete ppTempBuffers[i];
delete ppTempBuffers;
delete buffers;
}
}
@ -409,14 +702,14 @@ public:
{
if (!IsOpened()) return;
uint32 samples = ms2samples(milliseconds);
int32 block = samples / NUM_VAG_SAMPLES_IN_BLOCK;
uint32 block = samples / NUM_VAG_SAMPLES_IN_BLOCK;
if (block > m_nNumberOfBlocks)
{
samples = 0;
block = 0;
}
if (block != m_CurrentBlock)
ReadBlock(block);
m_bBlockRead = false;
uint32 remainingSamples = samples - block * NUM_VAG_SAMPLES_IN_BLOCK;
uint32 newLine = remainingSamples / VAG_SAMPLES_IN_LINE / VAG_LINE_SIZE;
@ -425,7 +718,7 @@ public:
{
m_CurrentBlock = block;
m_LineInBlock = newLine;
for (int i = 0; i < GetChannels(); i++)
for (uint32 i = 0; i < GetChannels(); i++)
decoders[i].ResetState();
}
@ -448,18 +741,19 @@ public:
if (m_CurrentBlock == m_nNumberOfBlocks) return 0;
int size = 0;
int numberOfRequiredLines = GetBufferSamples() / GetChannels() / VAG_SAMPLES_IN_LINE;
int numberOfRequiredLines = GetBufferSamples() / m_nChannels / VAG_SAMPLES_IN_LINE;
int numberOfRemainingLines = (m_nNumberOfBlocks - m_CurrentBlock) * NUM_VAG_LINES_IN_BLOCK - m_LineInBlock;
int bufSizePerChannel = Min(numberOfRequiredLines, numberOfRemainingLines) * VAG_SAMPLES_IN_LINE * GetSampleSize();
if (numberOfRequiredLines > numberOfRemainingLines)
numberOfRemainingLines = numberOfRemainingLines;
int16* buffers[2] = { (int16*)buffer, &((int16*)buffer)[bufSizePerChannel / GetSampleSize()] };
for (uint32 i = 0; i < m_nChannels; i++)
buffers[i] = (int16*)((int8*)buffer + bufSizePerChannel * i);
while (size < bufSizePerChannel)
{
for (int i = 0; i < GetChannels(); i++)
for (uint32 i = 0; i < m_nChannels; i++)
{
decoders[i].Decode(ppTempBuffers[i] + m_LineInBlock * VAG_LINE_SIZE, buffers[i], VAG_LINE_SIZE);
buffers[i] += VAG_SAMPLES_IN_LINE;
@ -476,7 +770,7 @@ public:
}
}
return bufSizePerChannel * GetChannels();
return bufSizePerChannel * m_nChannels;
}
};
#else
@ -621,7 +915,11 @@ CStream::CStream(char *filename, ALuint *sources, ALuint (&buffers)[NUM_STREAMBU
if (!strcasecmp(&m_aFilename[strlen(m_aFilename) - strlen(".mp3")], ".mp3"))
m_pSoundFile = new CMP3File(m_aFilename);
else if (!strcasecmp(&m_aFilename[strlen(m_aFilename) - strlen(".wav")], ".wav"))
#ifdef AUDIO_OAL_USE_SNDFILE
m_pSoundFile = new CSndFile(m_aFilename);
#else
m_pSoundFile = new CWavFile(m_aFilename);
#endif
else if (!strcasecmp(&m_aFilename[strlen(m_aFilename) - strlen(".vb")], ".VB"))
m_pSoundFile = new CVbFile(m_aFilename, overrideSampleRate);
#else
@ -922,12 +1220,15 @@ void CStream::Update()
// Relying a lot on left buffer states in here
do
{
//alSourcef(m_pAlSources[0], AL_ROLLOFF_FACTOR, 0.0f);
alGetSourcei(m_pAlSources[0], AL_SOURCE_STATE, &sourceState[0]);
alGetSourcei(m_pAlSources[0], AL_BUFFERS_PROCESSED, &buffersProcessed[0]);
//alSourcef(m_pAlSources[1], AL_ROLLOFF_FACTOR, 0.0f);
alGetSourcei(m_pAlSources[1], AL_SOURCE_STATE, &sourceState[1]);
alGetSourcei(m_pAlSources[1], AL_BUFFERS_PROCESSED, &buffersProcessed[1]);
} while (buffersProcessed[0] != buffersProcessed[1]);
ALint looping = AL_FALSE;
alGetSourcei(m_pAlSources[0], AL_LOOPING, &looping);

View File

@ -352,6 +352,7 @@ enum Config {
#define RADIO_SCROLL_TO_PREV_STATION
#define AUDIO_CACHE
//#define PS2_AUDIO // changes audio paths for cutscenes and radio to PS2 paths, needs vbdec to support VB with MSS
//#define AUDIO_OAL_USE_SNDFILE // use libsndfile to decode WAVs instead of our internal decoder
// IMG
#define BIG_IMG // allows to read larger img files