mirror of https://git.rip/DMCA_FUCKER/re3.git
Implementing our own WAV decoder to replace SndFile
This commit is contained in:
Sergeanur
parent
a6409fb445
commit
493f6cb578
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@ -8,10 +8,14 @@
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#include <opusfile.h>
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#else
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#ifdef _WIN32
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#ifdef AUDIO_OAL_USE_SNDFILE
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#pragma comment( lib, "libsndfile-1.lib" )
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#endif
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#pragma comment( lib, "libmpg123-0.lib" )
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#endif
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#ifdef AUDIO_OAL_USE_SNDFILE
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#include <sndfile.h>
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#endif
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#include <mpg123.h>
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#endif
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@ -78,6 +82,290 @@ public:
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CSortStereoBuffer SortStereoBuffer;
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#ifndef AUDIO_OPUS
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class CImaADPCMDecoder
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{
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const uint16 StepTable[89] = {
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7, 8, 9, 10, 11, 12, 13, 14,
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16, 17, 19, 21, 23, 25, 28, 31,
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34, 37, 41, 45, 50, 55, 60, 66,
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73, 80, 88, 97, 107, 118, 130, 143,
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157, 173, 190, 209, 230, 253, 279, 307,
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337, 371, 408, 449, 494, 544, 598, 658,
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724, 796, 876, 963, 1060, 1166, 1282, 1411,
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1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024,
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3327, 3660, 4026, 4428, 4871, 5358, 5894, 6484,
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7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
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15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794,
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32767
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};
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int16 Sample, StepIndex;
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public:
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CImaADPCMDecoder()
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{
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Init(0, 0);
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}
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void Init(int16 _Sample, int16 _StepIndex)
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{
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Sample = _Sample;
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StepIndex = _StepIndex;
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}
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void Decode(uint8 *inbuf, int16 *_outbuf, size_t size)
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{
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int16* outbuf = _outbuf;
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for (size_t i = 0; i < size; i++)
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{
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*(outbuf++) = DecodeSample(inbuf[i] & 0xF);
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*(outbuf++) = DecodeSample(inbuf[i] >> 4);
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}
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}
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int16 DecodeSample(uint8 adpcm)
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{
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uint16 step = StepTable[StepIndex];
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if (adpcm & 4)
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StepIndex += ((adpcm & 3) + 1) * 2;
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else
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StepIndex--;
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StepIndex = clamp(StepIndex, 0, 88);
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int delta = step >> 3;
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if (adpcm & 1) delta += step >> 2;
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if (adpcm & 2) delta += step >> 1;
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if (adpcm & 4) delta += step;
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if (adpcm & 8) delta = -delta;
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int newSample = Sample + delta;
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Sample = clamp(newSample, -32768, 32767);
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return Sample;
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}
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};
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class CWavFile : public IDecoder
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{
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enum
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{
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WAVEFMT_PCM = 1,
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WAVEFMT_IMA_ADPCM = 0x11,
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WAVEFMT_XBOX_ADPCM = 0x69,
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};
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struct tDataHeader
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{
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uint32 ID;
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uint32 Size;
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};
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struct tFormatHeader
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{
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uint16 AudioFormat;
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uint16 NumChannels;
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uint32 SampleRate;
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uint32 ByteRate;
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uint16 BlockAlign;
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uint16 BitsPerSample;
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uint16 extra[2]; // adpcm only
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tFormatHeader() { memset(this, 0, sizeof(*this)); }
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};
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FILE* pFile;
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bool bIsOpen;
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tFormatHeader FormatHeader;
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uint32 DataStartOffset;
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uint32 SampleCount;
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uint32 SamplesPerBlock;
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// ADPCM things
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uint8 *AdpcmBlock;
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int16 **buffers;
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CImaADPCMDecoder* decoders;
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void Close()
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{
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if (pFile) {
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fclose(pFile);
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pFile = nil;
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}
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if (AdpcmBlock) delete AdpcmBlock;
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if (buffers) delete buffers;
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if (decoders) delete decoders;
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}
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public:
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CWavFile(const char* path) : bIsOpen(false), DataStartOffset(0), SampleCount(0), SamplesPerBlock(0), AdpcmBlock(nil), buffers(nil), decoders(nil)
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{
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pFile = fopen(path, "rb");
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if (!pFile) return;
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#define CLOSE_ON_ERROR(op)\
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if (op) { \
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Close(); \
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return; \
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}
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tDataHeader DataHeader;
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CLOSE_ON_ERROR(fread(&DataHeader, sizeof(DataHeader), 1, pFile) == 0);
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CLOSE_ON_ERROR(DataHeader.ID != 'FFIR');
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int WAVE;
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CLOSE_ON_ERROR(fread(&WAVE, 4, 1, pFile) == 0);
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CLOSE_ON_ERROR(WAVE != 'EVAW')
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CLOSE_ON_ERROR(fread(&DataHeader, sizeof(DataHeader), 1, pFile) == 0);
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CLOSE_ON_ERROR(DataHeader.ID != ' tmf');
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CLOSE_ON_ERROR(fread(&FormatHeader, Min(DataHeader.Size, sizeof(tFormatHeader)), 1, pFile) == 0);
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CLOSE_ON_ERROR(DataHeader.Size > sizeof(tFormatHeader));
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switch (FormatHeader.AudioFormat)
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{
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case WAVEFMT_XBOX_ADPCM:
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FormatHeader.AudioFormat = WAVEFMT_IMA_ADPCM;
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case WAVEFMT_IMA_ADPCM:
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SamplesPerBlock = (FormatHeader.BlockAlign / FormatHeader.NumChannels - 4) * 2 + 1;
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AdpcmBlock = new uint8[FormatHeader.BlockAlign];
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buffers = new int16*[FormatHeader.NumChannels];
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decoders = new CImaADPCMDecoder[FormatHeader.NumChannels];
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break;
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case WAVEFMT_PCM:
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SamplesPerBlock = 1;
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if (FormatHeader.BitsPerSample != 16)
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{
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debug("Unsupported PCM (%d bits), only signed 16-bit is supported (%s)\n", FormatHeader.BitsPerSample, path);
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return;
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}
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break;
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default:
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debug("Unsupported wav format 0x%x (%s)\n", FormatHeader.AudioFormat, path);
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return;
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}
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while (true) {
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CLOSE_ON_ERROR(fread(&DataHeader, sizeof(DataHeader), 1, pFile) == 0);
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if (DataHeader.ID == 'atad')
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break;
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fseek(pFile, DataHeader.Size, SEEK_CUR);
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}
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DataStartOffset = ftell(pFile);
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SampleCount = DataHeader.Size / FormatHeader.BlockAlign * SamplesPerBlock;
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bIsOpen = true;
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#undef CLOSE_ON_ERROR
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}
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~CWavFile()
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{
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Close();
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}
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bool IsOpened()
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{
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return bIsOpen;
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}
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uint32 GetSampleSize()
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{
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return sizeof(uint16);
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}
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uint32 GetSampleCount()
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{
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return SampleCount;
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}
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uint32 GetSampleRate()
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{
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return FormatHeader.SampleRate;
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}
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uint32 GetChannels()
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{
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return FormatHeader.NumChannels;
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}
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void Seek(uint32 milliseconds)
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{
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if (!IsOpened()) return;
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fseek(pFile, DataStartOffset + ms2samples(milliseconds) / SamplesPerBlock * FormatHeader.BlockAlign, SEEK_SET);
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}
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uint32 Tell()
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{
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if (!IsOpened()) return 0;
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return samples2ms((ftell(pFile) - DataStartOffset) / FormatHeader.BlockAlign * SamplesPerBlock);
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}
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#define SAMPLES_IN_LINE (8)
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uint32 Decode(void* buffer)
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{
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if (!IsOpened()) return 0;
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if (FormatHeader.AudioFormat == WAVEFMT_PCM)
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{
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uint32 size = fread(buffer, 1, GetBufferSize(), pFile);
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if (FormatHeader.NumChannels == 2)
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SortStereoBuffer.SortStereo(buffer, size);
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return size;
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}
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else if (FormatHeader.AudioFormat == WAVEFMT_IMA_ADPCM)
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{
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uint32 MaxSamples = GetBufferSamples() / FormatHeader.NumChannels;
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uint32 CurSample = (ftell(pFile) - DataStartOffset) / FormatHeader.BlockAlign * SamplesPerBlock;
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MaxSamples = Min(MaxSamples, SampleCount - CurSample);
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MaxSamples = MaxSamples / SamplesPerBlock * SamplesPerBlock;
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uint32 OutBufSizePerChannel = MaxSamples * GetSampleSize();
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uint32 OutBufSize = OutBufSizePerChannel * FormatHeader.NumChannels;
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int16** buffers = new int16*[FormatHeader.NumChannels];
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CImaADPCMDecoder* decoders = new CImaADPCMDecoder[FormatHeader.NumChannels];
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for (uint32 i = 0; i < FormatHeader.NumChannels; i++)
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buffers[i] = (int16*)((int8*)buffer + OutBufSizePerChannel * i);
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uint32 samplesRead = 0;
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while (samplesRead < MaxSamples)
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{
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uint8* AdpcmBuf = AdpcmBlock;
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if (fread(AdpcmBlock, 1, FormatHeader.BlockAlign, pFile) == 0)
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return 0;
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for (uint32 i = 0; i < FormatHeader.NumChannels; i++)
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{
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int16 Sample = *(int16*)AdpcmBuf;
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AdpcmBuf += sizeof(int16);
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int16 Step = *(int16*)AdpcmBuf;
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AdpcmBuf += sizeof(int16);
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decoders[i].Init(Sample, Step);
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*(buffers[i]) = Sample;
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buffers[i]++;
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}
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samplesRead++;
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for (uint32 s = 1; s < SamplesPerBlock; s += SAMPLES_IN_LINE)
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{
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for (uint32 i = 0; i < FormatHeader.NumChannels; i++)
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{
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decoders[i].Decode(AdpcmBuf, buffers[i], SAMPLES_IN_LINE / 2);
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AdpcmBuf += SAMPLES_IN_LINE / 2;
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buffers[i] += SAMPLES_IN_LINE;
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}
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samplesRead += SAMPLES_IN_LINE;
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}
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}
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return OutBufSize;
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}
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return 0;
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}
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};
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#ifdef AUDIO_OAL_USE_SNDFILE
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class CSndFile : public IDecoder
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{
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SNDFILE *m_pfSound;
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@ -146,6 +434,7 @@ public:
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return size;
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}
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};
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#endif
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#ifdef _WIN32
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// fuzzy seek eliminates stutter when playing ADF but spams errors a lot (nothing breaks though)
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@ -280,7 +569,7 @@ public:
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static short quantize(double sample)
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{
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int a = int(sample + 0.5);
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return short(clamp(int(sample + 0.5), -32768, 32767));
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return short(clamp(a, -32768, 32767));
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}
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void Decode(void* _inbuf, int16* _outbuf, size_t size)
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@ -336,6 +625,7 @@ class CVbFile : public IDecoder
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size_t m_CurrentBlock;
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uint8** ppTempBuffers;
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int16** buffers;
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void ReadBlock(int32 block = -1)
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{
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@ -349,22 +639,24 @@ class CVbFile : public IDecoder
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}
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public:
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CVbFile(const char* path, uint32 nSampleRate = 32000, uint8 nChannels = 2) : m_nSampleRate(nSampleRate), m_nChannels(nChannels)
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CVbFile(const char* path, uint32 nSampleRate = 32000, uint8 nChannels = 2) : m_nSampleRate(nSampleRate), m_nChannels(nChannels), decoders(nil), ppTempBuffers(nil), buffers(nil),
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m_FileSize(0), m_nNumberOfBlocks(0), m_bBlockRead(false), m_LineInBlock(0), m_CurrentBlock(0)
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{
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pFile = fopen(path, "rb");
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if (pFile) {
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fseek(pFile, 0, SEEK_END);
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m_FileSize = ftell(pFile);
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fseek(pFile, 0, SEEK_SET);
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m_nNumberOfBlocks = m_FileSize / (nChannels * VB_BLOCK_SIZE);
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decoders = new CVagDecoder[nChannels];
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m_CurrentBlock = 0;
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m_LineInBlock = 0;
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m_bBlockRead = false;
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ppTempBuffers = new uint8 * [nChannels];
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for (uint8 i = 0; i < nChannels; i++)
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ppTempBuffers[i] = new uint8[VB_BLOCK_SIZE];
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}
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if (!pFile) return;
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fseek(pFile, 0, SEEK_END);
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m_FileSize = ftell(pFile);
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fseek(pFile, 0, SEEK_SET);
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m_nNumberOfBlocks = m_FileSize / (nChannels * VB_BLOCK_SIZE);
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decoders = new CVagDecoder[nChannels];
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m_CurrentBlock = 0;
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m_LineInBlock = 0;
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m_bBlockRead = false;
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ppTempBuffers = new uint8*[nChannels];
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buffers = new int16*[nChannels];
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for (uint8 i = 0; i < nChannels; i++)
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ppTempBuffers[i] = new uint8[VB_BLOCK_SIZE];
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}
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~CVbFile()
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@ -376,6 +668,7 @@ public:
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for (int i = 0; i < m_nChannels; i++)
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delete ppTempBuffers[i];
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delete ppTempBuffers;
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delete buffers;
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}
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}
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@ -409,14 +702,14 @@ public:
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{
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if (!IsOpened()) return;
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uint32 samples = ms2samples(milliseconds);
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int32 block = samples / NUM_VAG_SAMPLES_IN_BLOCK;
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uint32 block = samples / NUM_VAG_SAMPLES_IN_BLOCK;
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if (block > m_nNumberOfBlocks)
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{
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samples = 0;
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block = 0;
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}
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if (block != m_CurrentBlock)
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ReadBlock(block);
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m_bBlockRead = false;
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uint32 remainingSamples = samples - block * NUM_VAG_SAMPLES_IN_BLOCK;
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uint32 newLine = remainingSamples / VAG_SAMPLES_IN_LINE / VAG_LINE_SIZE;
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@ -425,7 +718,7 @@ public:
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{
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m_CurrentBlock = block;
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m_LineInBlock = newLine;
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for (int i = 0; i < GetChannels(); i++)
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for (uint32 i = 0; i < GetChannels(); i++)
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decoders[i].ResetState();
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}
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@ -448,18 +741,19 @@ public:
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if (m_CurrentBlock == m_nNumberOfBlocks) return 0;
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int size = 0;
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int numberOfRequiredLines = GetBufferSamples() / GetChannels() / VAG_SAMPLES_IN_LINE;
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int numberOfRequiredLines = GetBufferSamples() / m_nChannels / VAG_SAMPLES_IN_LINE;
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int numberOfRemainingLines = (m_nNumberOfBlocks - m_CurrentBlock) * NUM_VAG_LINES_IN_BLOCK - m_LineInBlock;
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int bufSizePerChannel = Min(numberOfRequiredLines, numberOfRemainingLines) * VAG_SAMPLES_IN_LINE * GetSampleSize();
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if (numberOfRequiredLines > numberOfRemainingLines)
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numberOfRemainingLines = numberOfRemainingLines;
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int16* buffers[2] = { (int16*)buffer, &((int16*)buffer)[bufSizePerChannel / GetSampleSize()] };
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for (uint32 i = 0; i < m_nChannels; i++)
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buffers[i] = (int16*)((int8*)buffer + bufSizePerChannel * i);
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while (size < bufSizePerChannel)
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{
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for (int i = 0; i < GetChannels(); i++)
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for (uint32 i = 0; i < m_nChannels; i++)
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{
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decoders[i].Decode(ppTempBuffers[i] + m_LineInBlock * VAG_LINE_SIZE, buffers[i], VAG_LINE_SIZE);
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buffers[i] += VAG_SAMPLES_IN_LINE;
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@ -476,7 +770,7 @@ public:
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}
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}
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return bufSizePerChannel * GetChannels();
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return bufSizePerChannel * m_nChannels;
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}
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};
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#else
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@ -621,7 +915,11 @@ CStream::CStream(char *filename, ALuint *sources, ALuint (&buffers)[NUM_STREAMBU
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if (!strcasecmp(&m_aFilename[strlen(m_aFilename) - strlen(".mp3")], ".mp3"))
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m_pSoundFile = new CMP3File(m_aFilename);
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else if (!strcasecmp(&m_aFilename[strlen(m_aFilename) - strlen(".wav")], ".wav"))
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#ifdef AUDIO_OAL_USE_SNDFILE
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m_pSoundFile = new CSndFile(m_aFilename);
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#else
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m_pSoundFile = new CWavFile(m_aFilename);
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#endif
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else if (!strcasecmp(&m_aFilename[strlen(m_aFilename) - strlen(".vb")], ".VB"))
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m_pSoundFile = new CVbFile(m_aFilename, overrideSampleRate);
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#else
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@ -922,12 +1220,15 @@ void CStream::Update()
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// Relying a lot on left buffer states in here
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//alSourcef(m_pAlSources[0], AL_ROLLOFF_FACTOR, 0.0f);
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alGetSourcei(m_pAlSources[0], AL_SOURCE_STATE, &sourceState[0]);
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alGetSourcei(m_pAlSources[0], AL_BUFFERS_PROCESSED, &buffersProcessed[0]);
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//alSourcef(m_pAlSources[1], AL_ROLLOFF_FACTOR, 0.0f);
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alGetSourcei(m_pAlSources[1], AL_SOURCE_STATE, &sourceState[1]);
|
||||
alGetSourcei(m_pAlSources[1], AL_BUFFERS_PROCESSED, &buffersProcessed[1]);
|
||||
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);
|
||||
|
|
|
|||
|
|
@ -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
|
||||
|
|
|
|||
Loading…
Reference in New Issue