mirror of
https://git.rip/DMCA_FUCKER/re3.git
synced 2024-11-13 06:19:14 +00:00
1386 lines
30 KiB
C++
1386 lines
30 KiB
C++
#include "common.h"
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#ifdef AUDIO_OAL
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#include "stream.h"
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#include "sampman.h"
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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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#ifdef AUDIO_OAL_USE_MPG123
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#pragma comment( lib, "libmpg123-0.lib" )
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#endif
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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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#ifdef AUDIO_OAL_USE_MPG123
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#include <mpg123.h>
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#endif
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#ifdef AUDIO_OAL_USE_OPUS
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#include <opusfile.h>
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#endif
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#ifndef _WIN32
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#include "crossplatform.h"
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#endif
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/*
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As we ran onto an issue of having different volume levels for mono streams
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and stereo streams we are now handling all the stereo panning ourselves.
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Each stream now has two sources - one panned to the left and one to the right,
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and uses two separate buffers to store data for each individual channel.
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For that we also have to reshuffle all decoded PCM stereo data from LRLRLRLR to
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LLLLRRRR (handled by CSortStereoBuffer).
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*/
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class CSortStereoBuffer
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{
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uint16* PcmBuf;
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size_t BufSize;
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public:
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CSortStereoBuffer() : PcmBuf(nil), BufSize(0) {}
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~CSortStereoBuffer()
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{
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if (PcmBuf)
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free(PcmBuf);
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}
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uint16* GetBuffer(size_t size)
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{
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if (size == 0) return nil;
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if (!PcmBuf)
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{
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BufSize = size;
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PcmBuf = (uint16*)malloc(BufSize);
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}
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else if (BufSize < size)
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{
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BufSize = size;
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PcmBuf = (uint16*)realloc(PcmBuf, size);
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}
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return PcmBuf;
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}
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void SortStereo(void* buf, size_t size)
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{
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uint16* InBuf = (uint16*)buf;
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uint16* OutBuf = GetBuffer(size);
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if (!OutBuf) return;
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size_t rightStart = size / 4;
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for (size_t i = 0; i < size / 4; i++)
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{
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OutBuf[i] = InBuf[i*2];
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OutBuf[i+rightStart] = InBuf[i*2+1];
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}
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memcpy(InBuf, OutBuf, size);
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}
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};
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CSortStereoBuffer SortStereoBuffer;
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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 *m_pFile;
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bool m_bIsOpen;
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tFormatHeader m_FormatHeader;
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uint32 m_DataStartOffset; // TODO: 64 bit?
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uint32 m_nSampleCount;
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uint32 m_nSamplesPerBlock;
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// ADPCM things
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uint8 *m_pAdpcmBuffer;
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int16 **m_ppPcmBuffers;
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CImaADPCMDecoder *m_pAdpcmDecoders;
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void Close()
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{
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if (m_pFile) {
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fclose(m_pFile);
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m_pFile = nil;
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}
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delete[] m_pAdpcmBuffer;
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delete[] m_ppPcmBuffers;
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delete[] m_pAdpcmDecoders;
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}
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uint32 GetCurrentSample() const
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{
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// TODO: 64 bit?
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uint32 FilePos = ftell(m_pFile);
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if (FilePos <= m_DataStartOffset)
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return 0;
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return (FilePos - m_DataStartOffset) / m_FormatHeader.BlockAlign * m_nSamplesPerBlock;
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}
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public:
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CWavFile(const char* path) : m_bIsOpen(false), m_DataStartOffset(0), m_nSampleCount(0), m_nSamplesPerBlock(0), m_pAdpcmBuffer(nil), m_ppPcmBuffers(nil), m_pAdpcmDecoders(nil)
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{
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m_pFile = fopen(path, "rb");
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if (!m_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, m_pFile) == 0);
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CLOSE_ON_ERROR(DataHeader.ID != 'FFIR');
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// TODO? validate filesizes
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int WAVE;
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CLOSE_ON_ERROR(fread(&WAVE, 4, 1, m_pFile) == 0);
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CLOSE_ON_ERROR(WAVE != 'EVAW')
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CLOSE_ON_ERROR(fread(&DataHeader, sizeof(DataHeader), 1, m_pFile) == 0);
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CLOSE_ON_ERROR(DataHeader.ID != ' tmf');
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CLOSE_ON_ERROR(fread(&m_FormatHeader, Min(DataHeader.Size, sizeof(tFormatHeader)), 1, m_pFile) == 0);
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CLOSE_ON_ERROR(DataHeader.Size > sizeof(tFormatHeader));
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switch (m_FormatHeader.AudioFormat)
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{
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case WAVEFMT_XBOX_ADPCM:
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m_FormatHeader.AudioFormat = WAVEFMT_IMA_ADPCM;
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case WAVEFMT_IMA_ADPCM:
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m_nSamplesPerBlock = (m_FormatHeader.BlockAlign / m_FormatHeader.NumChannels - 4) * 2 + 1;
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m_pAdpcmBuffer = new uint8[m_FormatHeader.BlockAlign];
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m_ppPcmBuffers = new int16*[m_FormatHeader.NumChannels];
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m_pAdpcmDecoders = new CImaADPCMDecoder[m_FormatHeader.NumChannels];
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break;
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case WAVEFMT_PCM:
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m_nSamplesPerBlock = 1;
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if (m_FormatHeader.BitsPerSample != 16)
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{
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debug("Unsupported PCM (%d bits), only signed 16-bit is supported (%s)\n", m_FormatHeader.BitsPerSample, path);
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Close();
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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", m_FormatHeader.AudioFormat, path);
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Close();
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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, m_pFile) == 0);
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if (DataHeader.ID == 'atad')
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break;
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fseek(m_pFile, DataHeader.Size, SEEK_CUR);
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// TODO? validate data size
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// maybe check if there no extreme custom headers that might break this
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}
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m_DataStartOffset = ftell(m_pFile);
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m_nSampleCount = DataHeader.Size / m_FormatHeader.BlockAlign * m_nSamplesPerBlock;
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m_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 m_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 m_nSampleCount;
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}
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uint32 GetSampleRate()
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{
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return m_FormatHeader.SampleRate;
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}
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uint32 GetChannels()
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{
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return m_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(m_pFile, m_DataStartOffset + ms2samples(milliseconds) / m_nSamplesPerBlock * m_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(GetCurrentSample());
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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 (m_FormatHeader.AudioFormat == WAVEFMT_PCM)
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{
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// just read the file and sort the samples
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uint32 size = fread(buffer, 1, GetBufferSize(), m_pFile);
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if (m_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 (m_FormatHeader.AudioFormat == WAVEFMT_IMA_ADPCM)
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{
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// trim the buffer size if we're at the end of our file
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uint32 nMaxSamples = GetBufferSamples() / m_FormatHeader.NumChannels;
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uint32 nSamplesLeft = m_nSampleCount - GetCurrentSample();
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nMaxSamples = Min(nMaxSamples, nSamplesLeft);
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// align sample count to our block
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nMaxSamples = nMaxSamples / m_nSamplesPerBlock * m_nSamplesPerBlock;
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// count the size of output buffer
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uint32 OutBufSizePerChannel = nMaxSamples * GetSampleSize();
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uint32 OutBufSize = OutBufSizePerChannel * m_FormatHeader.NumChannels;
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// calculate the pointers to individual channel buffers
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for (uint32 i = 0; i < m_FormatHeader.NumChannels; i++)
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m_ppPcmBuffers[i] = (int16*)((int8*)buffer + OutBufSizePerChannel * i);
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uint32 samplesRead = 0;
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while (samplesRead < nMaxSamples)
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{
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// read the file
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uint8 *pAdpcmBuf = m_pAdpcmBuffer;
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if (fread(m_pAdpcmBuffer, 1, m_FormatHeader.BlockAlign, m_pFile) == 0)
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return 0;
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// get the first sample in adpcm block and initialise the decoder(s)
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for (uint32 i = 0; i < m_FormatHeader.NumChannels; i++)
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{
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int16 Sample = *(int16*)pAdpcmBuf;
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pAdpcmBuf += sizeof(int16);
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int16 Step = *(int16*)pAdpcmBuf;
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pAdpcmBuf += sizeof(int16);
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m_pAdpcmDecoders[i].Init(Sample, Step);
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*(m_ppPcmBuffers[i]) = Sample;
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m_ppPcmBuffers[i]++;
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}
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samplesRead++;
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// decode the rest of the block
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for (uint32 s = 1; s < m_nSamplesPerBlock; s += SAMPLES_IN_LINE)
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{
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for (uint32 i = 0; i < m_FormatHeader.NumChannels; i++)
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{
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m_pAdpcmDecoders[i].Decode(pAdpcmBuf, m_ppPcmBuffers[i], SAMPLES_IN_LINE / 2);
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pAdpcmBuf += SAMPLES_IN_LINE / 2;
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m_ppPcmBuffers[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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SF_INFO m_soundInfo;
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public:
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CSndFile(const char *path) :
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m_pfSound(nil)
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{
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memset(&m_soundInfo, 0, sizeof(m_soundInfo));
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m_pfSound = sf_open(path, SFM_READ, &m_soundInfo);
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}
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~CSndFile()
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{
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if ( m_pfSound )
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{
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sf_close(m_pfSound);
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m_pfSound = nil;
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}
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}
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bool IsOpened()
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{
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return m_pfSound != nil;
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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 m_soundInfo.frames;
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}
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uint32 GetSampleRate()
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{
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return m_soundInfo.samplerate;
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}
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uint32 GetChannels()
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{
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return m_soundInfo.channels;
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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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sf_seek(m_pfSound, ms2samples(milliseconds), SF_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(sf_seek(m_pfSound, 0, SF_SEEK_CUR));
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}
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uint32 Decode(void *buffer)
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{
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if ( !IsOpened() ) return 0;
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size_t size = sf_read_short(m_pfSound, (short*)buffer, GetBufferSamples()) * GetSampleSize();
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if (GetChannels()==2)
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SortStereoBuffer.SortStereo(buffer, size);
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return size;
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}
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};
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#endif
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#ifdef AUDIO_OAL_USE_MPG123
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// fuzzy seek eliminates stutter when playing ADF but spams errors a lot (nothing breaks though)
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#define MP3_USE_FUZZY_SEEK
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class CMP3File : public IDecoder
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{
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protected:
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mpg123_handle *m_pMH;
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bool m_bOpened;
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uint32 m_nRate;
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uint32 m_nChannels;
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CMP3File() :
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m_pMH(nil),
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m_bOpened(false),
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m_nRate(0),
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m_nChannels(0) {}
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public:
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CMP3File(const char *path) :
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m_pMH(nil),
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m_bOpened(false),
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m_nRate(0),
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m_nChannels(0)
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{
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m_pMH = mpg123_new(nil, nil);
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if ( m_pMH )
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{
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#ifdef MP3_USE_FUZZY_SEEK
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mpg123_param(m_pMH, MPG123_FLAGS, MPG123_FUZZY | MPG123_SEEKBUFFER | MPG123_GAPLESS | MPG123_QUIET, 0.0);
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#endif
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long rate = 0;
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int channels = 0;
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int encoding = 0;
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m_bOpened = mpg123_open(m_pMH, path) == MPG123_OK
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&& mpg123_getformat(m_pMH, &rate, &channels, &encoding) == MPG123_OK;
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m_nRate = rate;
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m_nChannels = channels;
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if ( IsOpened() )
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{
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mpg123_format_none(m_pMH);
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mpg123_format(m_pMH, rate, channels, encoding);
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}
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}
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}
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~CMP3File()
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{
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if ( m_pMH )
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{
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mpg123_close(m_pMH);
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mpg123_delete(m_pMH);
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m_pMH = nil;
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}
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}
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bool IsOpened()
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{
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return m_bOpened;
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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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if ( !IsOpened() ) return 0;
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return mpg123_length(m_pMH);
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}
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uint32 GetSampleRate()
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{
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return m_nRate;
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}
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uint32 GetChannels()
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{
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return m_nChannels;
|
|
}
|
|
|
|
void Seek(uint32 milliseconds)
|
|
{
|
|
if ( !IsOpened() ) return;
|
|
mpg123_seek(m_pMH, ms2samples(milliseconds), SEEK_SET);
|
|
}
|
|
|
|
uint32 Tell()
|
|
{
|
|
if ( !IsOpened() ) return 0;
|
|
return samples2ms(mpg123_tell(m_pMH));
|
|
}
|
|
|
|
uint32 Decode(void *buffer)
|
|
{
|
|
if ( !IsOpened() ) return 0;
|
|
|
|
size_t size;
|
|
int err = mpg123_read(m_pMH, (unsigned char *)buffer, GetBufferSize(), &size);
|
|
#if defined(__LP64__) || defined(_WIN64)
|
|
assert("We can't handle audio files more then 2 GB yet :shrug:" && (size < UINT32_MAX));
|
|
#endif
|
|
if (err != MPG123_OK && err != MPG123_DONE) return 0;
|
|
if (GetChannels() == 2)
|
|
SortStereoBuffer.SortStereo(buffer, size);
|
|
return (uint32)size;
|
|
}
|
|
};
|
|
|
|
class CADFFile : public CMP3File
|
|
{
|
|
static ssize_t r_read(void* fh, void* buf, size_t size)
|
|
{
|
|
size_t bytesRead = fread(buf, 1, size, (FILE*)fh);
|
|
uint8* _buf = (uint8*)buf;
|
|
for (size_t i = 0; i < size; i++)
|
|
_buf[i] ^= 0x22;
|
|
return bytesRead;
|
|
}
|
|
static off_t r_seek(void* fh, off_t pos, int seekType)
|
|
{
|
|
fseek((FILE*)fh, pos, seekType);
|
|
return ftell((FILE*)fh);
|
|
}
|
|
static void r_close(void* fh)
|
|
{
|
|
fclose((FILE*)fh);
|
|
}
|
|
public:
|
|
CADFFile(const char* path)
|
|
{
|
|
m_pMH = mpg123_new(nil, nil);
|
|
if (m_pMH)
|
|
{
|
|
#ifdef MP3_USE_FUZZY_SEEK
|
|
mpg123_param(m_pMH, MPG123_FLAGS, MPG123_FUZZY | MPG123_SEEKBUFFER | MPG123_GAPLESS | MPG123_QUIET, 0.0);
|
|
#endif
|
|
long rate = 0;
|
|
int channels = 0;
|
|
int encoding = 0;
|
|
|
|
FILE* f = fopen(path, "rb");
|
|
|
|
m_bOpened = mpg123_replace_reader_handle(m_pMH, r_read, r_seek, r_close) == MPG123_OK
|
|
&& mpg123_open_handle(m_pMH, f) == MPG123_OK && mpg123_getformat(m_pMH, &rate, &channels, &encoding) == MPG123_OK;
|
|
m_nRate = rate;
|
|
m_nChannels = channels;
|
|
|
|
if (IsOpened())
|
|
{
|
|
mpg123_format_none(m_pMH);
|
|
mpg123_format(m_pMH, rate, channels, encoding);
|
|
}
|
|
}
|
|
}
|
|
};
|
|
|
|
#endif
|
|
#define VAG_LINE_SIZE (0x10)
|
|
#define VAG_SAMPLES_IN_LINE (28)
|
|
|
|
class CVagDecoder
|
|
{
|
|
const double f[5][2] = { { 0.0, 0.0 },
|
|
{ 60.0 / 64.0, 0.0 },
|
|
{ 115.0 / 64.0, -52.0 / 64.0 },
|
|
{ 98.0 / 64.0, -55.0 / 64.0 },
|
|
{ 122.0 / 64.0, -60.0 / 64.0 } };
|
|
|
|
double s_1;
|
|
double s_2;
|
|
public:
|
|
CVagDecoder()
|
|
{
|
|
ResetState();
|
|
}
|
|
|
|
void ResetState()
|
|
{
|
|
s_1 = s_2 = 0.0;
|
|
}
|
|
|
|
static short quantize(double sample)
|
|
{
|
|
int a = int(sample + 0.5);
|
|
return short(clamp(a, -32768, 32767));
|
|
}
|
|
|
|
void Decode(void* _inbuf, int16* _outbuf, size_t size)
|
|
{
|
|
uint8* inbuf = (uint8*)_inbuf;
|
|
int16* outbuf = _outbuf;
|
|
size &= ~(VAG_LINE_SIZE - 1);
|
|
|
|
while (size > 0) {
|
|
double samples[VAG_SAMPLES_IN_LINE];
|
|
|
|
int predict_nr, shift_factor, flags;
|
|
predict_nr = *(inbuf++);
|
|
shift_factor = predict_nr & 0xf;
|
|
predict_nr >>= 4;
|
|
flags = *(inbuf++);
|
|
if (flags == 7) // TODO: ignore?
|
|
break;
|
|
for (int i = 0; i < VAG_SAMPLES_IN_LINE; i += 2) {
|
|
int d = *(inbuf++);
|
|
int16 s = int16((d & 0xf) << 12);
|
|
samples[i] = (double)(s >> shift_factor);
|
|
s = int16((d & 0xf0) << 8);
|
|
samples[i + 1] = (double)(s >> shift_factor);
|
|
}
|
|
|
|
for (int i = 0; i < VAG_SAMPLES_IN_LINE; i++) {
|
|
samples[i] = samples[i] + s_1 * f[predict_nr][0] + s_2 * f[predict_nr][1];
|
|
s_2 = s_1;
|
|
s_1 = samples[i];
|
|
*(outbuf++) = quantize(samples[i] + 0.5);
|
|
}
|
|
size -= VAG_LINE_SIZE;
|
|
}
|
|
}
|
|
};
|
|
|
|
#define VB_BLOCK_SIZE (0x2000)
|
|
#define NUM_VAG_LINES_IN_BLOCK (VB_BLOCK_SIZE / VAG_LINE_SIZE)
|
|
#define NUM_VAG_SAMPLES_IN_BLOCK (NUM_VAG_LINES_IN_BLOCK * VAG_SAMPLES_IN_LINE)
|
|
|
|
class CVbFile : public IDecoder
|
|
{
|
|
FILE *m_pFile;
|
|
CVagDecoder *m_pVagDecoders;
|
|
|
|
size_t m_FileSize;
|
|
size_t m_nNumberOfBlocks;
|
|
|
|
uint32 m_nSampleRate;
|
|
uint8 m_nChannels;
|
|
bool m_bBlockRead;
|
|
uint16 m_LineInBlock;
|
|
size_t m_CurrentBlock;
|
|
|
|
uint8 **m_ppVagBuffers; // buffers that cache actual ADPCM file data
|
|
int16 **m_ppPcmBuffers;
|
|
|
|
void ReadBlock(int32 block = -1)
|
|
{
|
|
// just read next block if -1
|
|
if (block != -1)
|
|
fseek(m_pFile, block * m_nChannels * VB_BLOCK_SIZE, SEEK_SET);
|
|
|
|
for (int i = 0; i < m_nChannels; i++)
|
|
fread(m_ppVagBuffers[i], VB_BLOCK_SIZE, 1, m_pFile);
|
|
m_bBlockRead = true;
|
|
}
|
|
|
|
public:
|
|
CVbFile(const char* path, uint32 nSampleRate = 32000, uint8 nChannels = 2) : m_nSampleRate(nSampleRate), m_nChannels(nChannels), m_pVagDecoders(nil), m_ppVagBuffers(nil), m_ppPcmBuffers(nil),
|
|
m_FileSize(0), m_nNumberOfBlocks(0), m_bBlockRead(false), m_LineInBlock(0), m_CurrentBlock(0)
|
|
{
|
|
m_pFile = fopen(path, "rb");
|
|
if (!m_pFile) return;
|
|
|
|
fseek(m_pFile, 0, SEEK_END);
|
|
m_FileSize = ftell(m_pFile);
|
|
fseek(m_pFile, 0, SEEK_SET);
|
|
|
|
m_nNumberOfBlocks = m_FileSize / (nChannels * VB_BLOCK_SIZE);
|
|
m_pVagDecoders = new CVagDecoder[nChannels];
|
|
m_ppVagBuffers = new uint8*[nChannels];
|
|
m_ppPcmBuffers = new int16*[nChannels];
|
|
for (uint8 i = 0; i < nChannels; i++)
|
|
m_ppVagBuffers[i] = new uint8[VB_BLOCK_SIZE];
|
|
}
|
|
|
|
~CVbFile()
|
|
{
|
|
if (m_pFile)
|
|
{
|
|
fclose(m_pFile);
|
|
|
|
delete[] m_pVagDecoders;
|
|
for (int i = 0; i < m_nChannels; i++)
|
|
delete[] m_ppVagBuffers[i];
|
|
delete[] m_ppVagBuffers;
|
|
delete[] m_ppPcmBuffers;
|
|
}
|
|
}
|
|
|
|
bool IsOpened()
|
|
{
|
|
return m_pFile != nil;
|
|
}
|
|
|
|
uint32 GetSampleSize()
|
|
{
|
|
return sizeof(uint16);
|
|
}
|
|
|
|
uint32 GetSampleCount()
|
|
{
|
|
if (!IsOpened()) return 0;
|
|
return m_nNumberOfBlocks * NUM_VAG_LINES_IN_BLOCK * VAG_SAMPLES_IN_LINE;
|
|
}
|
|
|
|
uint32 GetSampleRate()
|
|
{
|
|
return m_nSampleRate;
|
|
}
|
|
|
|
uint32 GetChannels()
|
|
{
|
|
return m_nChannels;
|
|
}
|
|
|
|
void Seek(uint32 milliseconds)
|
|
{
|
|
if (!IsOpened()) return;
|
|
uint32 samples = ms2samples(milliseconds);
|
|
|
|
// find the block of our sample
|
|
uint32 block = samples / NUM_VAG_SAMPLES_IN_BLOCK;
|
|
if (block > m_nNumberOfBlocks)
|
|
{
|
|
samples = 0;
|
|
block = 0;
|
|
}
|
|
if (block != m_CurrentBlock)
|
|
m_bBlockRead = false;
|
|
|
|
// find a line of our sample within our block
|
|
uint32 remainingSamples = samples - block * NUM_VAG_SAMPLES_IN_BLOCK;
|
|
uint32 newLine = remainingSamples / VAG_SAMPLES_IN_LINE / VAG_LINE_SIZE;
|
|
|
|
if (m_CurrentBlock != block || m_LineInBlock != newLine)
|
|
{
|
|
m_CurrentBlock = block;
|
|
m_LineInBlock = newLine;
|
|
for (uint32 i = 0; i < GetChannels(); i++)
|
|
m_pVagDecoders[i].ResetState();
|
|
}
|
|
|
|
}
|
|
|
|
uint32 Tell()
|
|
{
|
|
if (!IsOpened()) return 0;
|
|
uint32 pos = (m_CurrentBlock * NUM_VAG_LINES_IN_BLOCK + m_LineInBlock) * VAG_SAMPLES_IN_LINE;
|
|
return samples2ms(pos);
|
|
}
|
|
|
|
uint32 Decode(void* buffer)
|
|
{
|
|
if (!IsOpened()) return 0;
|
|
|
|
if (m_CurrentBlock >= m_nNumberOfBlocks) return 0;
|
|
|
|
// cache current ADPCM block
|
|
if (!m_bBlockRead)
|
|
ReadBlock(m_CurrentBlock);
|
|
|
|
// trim the buffer size if we're at the end of our file
|
|
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();
|
|
|
|
// calculate the pointers to individual channel buffers
|
|
for (uint32 i = 0; i < m_nChannels; i++)
|
|
m_ppPcmBuffers[i] = (int16*)((int8*)buffer + bufSizePerChannel * i);
|
|
|
|
int size = 0;
|
|
while (size < bufSizePerChannel)
|
|
{
|
|
// decode the VAG lines
|
|
for (uint32 i = 0; i < m_nChannels; i++)
|
|
{
|
|
m_pVagDecoders[i].Decode(m_ppVagBuffers[i] + m_LineInBlock * VAG_LINE_SIZE, m_ppPcmBuffers[i], VAG_LINE_SIZE);
|
|
m_ppPcmBuffers[i] += VAG_SAMPLES_IN_LINE;
|
|
}
|
|
size += VAG_SAMPLES_IN_LINE * GetSampleSize();
|
|
m_LineInBlock++;
|
|
|
|
// block is over, read the next block
|
|
if (m_LineInBlock >= NUM_VAG_LINES_IN_BLOCK)
|
|
{
|
|
m_CurrentBlock++;
|
|
if (m_CurrentBlock >= m_nNumberOfBlocks) // end of file
|
|
break;
|
|
m_LineInBlock = 0;
|
|
ReadBlock();
|
|
}
|
|
}
|
|
|
|
return bufSizePerChannel * m_nChannels;
|
|
}
|
|
};
|
|
#ifdef AUDIO_OAL_USE_OPUS
|
|
class COpusFile : public IDecoder
|
|
{
|
|
OggOpusFile *m_FileH;
|
|
bool m_bOpened;
|
|
uint32 m_nRate;
|
|
uint32 m_nChannels;
|
|
public:
|
|
COpusFile(const char *path) : m_FileH(nil),
|
|
m_bOpened(false),
|
|
m_nRate(0),
|
|
m_nChannels(0)
|
|
{
|
|
int ret;
|
|
m_FileH = op_open_file(path, &ret);
|
|
|
|
if (m_FileH) {
|
|
m_nChannels = op_head(m_FileH, 0)->channel_count;
|
|
m_nRate = 48000;
|
|
const OpusTags *tags = op_tags(m_FileH, 0);
|
|
for (int i = 0; i < tags->comments; i++) {
|
|
if (strncmp(tags->user_comments[i], "SAMPLERATE", sizeof("SAMPLERATE")-1) == 0)
|
|
{
|
|
sscanf(tags->user_comments[i], "SAMPLERATE=%i", &m_nRate);
|
|
break;
|
|
}
|
|
}
|
|
|
|
m_bOpened = true;
|
|
}
|
|
}
|
|
|
|
~COpusFile()
|
|
{
|
|
if (m_FileH)
|
|
{
|
|
op_free(m_FileH);
|
|
m_FileH = nil;
|
|
}
|
|
}
|
|
|
|
bool IsOpened()
|
|
{
|
|
return m_bOpened;
|
|
}
|
|
|
|
uint32 GetSampleSize()
|
|
{
|
|
return sizeof(uint16);
|
|
}
|
|
|
|
uint32 GetSampleCount()
|
|
{
|
|
if ( !IsOpened() ) return 0;
|
|
return op_pcm_total(m_FileH, 0);
|
|
}
|
|
|
|
uint32 GetSampleRate()
|
|
{
|
|
return m_nRate;
|
|
}
|
|
|
|
uint32 GetChannels()
|
|
{
|
|
return m_nChannels;
|
|
}
|
|
|
|
void Seek(uint32 milliseconds)
|
|
{
|
|
if ( !IsOpened() ) return;
|
|
op_pcm_seek(m_FileH, ms2samples(milliseconds) / GetChannels());
|
|
}
|
|
|
|
uint32 Tell()
|
|
{
|
|
if ( !IsOpened() ) return 0;
|
|
return samples2ms(op_pcm_tell(m_FileH) * GetChannels());
|
|
}
|
|
|
|
uint32 Decode(void *buffer)
|
|
{
|
|
if ( !IsOpened() ) return 0;
|
|
|
|
int size = op_read(m_FileH, (opus_int16 *)buffer, GetBufferSamples(), NULL);
|
|
|
|
if (size < 0)
|
|
return 0;
|
|
|
|
if (GetChannels() == 2)
|
|
SortStereoBuffer.SortStereo(buffer, size * m_nChannels * GetSampleSize());
|
|
|
|
return size * m_nChannels * GetSampleSize();
|
|
}
|
|
};
|
|
#endif
|
|
|
|
void CStream::Initialise()
|
|
{
|
|
#ifdef AUDIO_OAL_USE_MPG123
|
|
mpg123_init();
|
|
#endif
|
|
}
|
|
|
|
void CStream::Terminate()
|
|
{
|
|
#ifdef AUDIO_OAL_USE_MPG123
|
|
mpg123_exit();
|
|
#endif
|
|
}
|
|
|
|
CStream::CStream(char *filename, ALuint *sources, ALuint (&buffers)[NUM_STREAMBUFFERS], uint32 overrideSampleRate) :
|
|
m_pAlSources(sources),
|
|
m_alBuffers(buffers),
|
|
m_pBuffer(nil),
|
|
m_bPaused(false),
|
|
m_bActive(false),
|
|
m_pSoundFile(nil),
|
|
m_bReset(false),
|
|
m_nVolume(0),
|
|
m_nPan(0),
|
|
m_nPosBeforeReset(0)
|
|
|
|
{
|
|
// Be case-insensitive on linux (from https://github.com/OneSadCookie/fcaseopen/)
|
|
#if !defined(_WIN32)
|
|
char *real = casepath(filename);
|
|
if (real) {
|
|
strcpy(m_aFilename, real);
|
|
free(real);
|
|
} else {
|
|
#else
|
|
{
|
|
#endif
|
|
strcpy(m_aFilename, filename);
|
|
}
|
|
|
|
DEV("Stream %s\n", m_aFilename);
|
|
|
|
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
|
|
#ifdef AUDIO_OAL_USE_MPG123
|
|
else 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(".adf")], ".adf"))
|
|
m_pSoundFile = new CADFFile(m_aFilename);
|
|
#endif
|
|
else if (!strcasecmp(&m_aFilename[strlen(m_aFilename) - strlen(".vb")], ".VB"))
|
|
m_pSoundFile = new CVbFile(m_aFilename, overrideSampleRate);
|
|
#ifdef AUDIO_OAL_USE_OPUS
|
|
else if (!strcasecmp(&m_aFilename[strlen(m_aFilename) - strlen(".opus")], ".opus"))
|
|
m_pSoundFile = new COpusFile(m_aFilename);
|
|
#endif
|
|
else
|
|
m_pSoundFile = nil;
|
|
|
|
if ( IsOpened() )
|
|
{
|
|
m_pBuffer = malloc(m_pSoundFile->GetBufferSize());
|
|
ASSERT(m_pBuffer!=nil);
|
|
|
|
DEV("AvgSamplesPerSec: %d\n", m_pSoundFile->GetAvgSamplesPerSec());
|
|
DEV("SampleCount: %d\n", m_pSoundFile->GetSampleCount());
|
|
DEV("SampleRate: %d\n", m_pSoundFile->GetSampleRate());
|
|
DEV("Channels: %d\n", m_pSoundFile->GetChannels());
|
|
DEV("Buffer Samples: %d\n", m_pSoundFile->GetBufferSamples());
|
|
DEV("Buffer sec: %f\n", (float(m_pSoundFile->GetBufferSamples()) / float(m_pSoundFile->GetChannels())/ float(m_pSoundFile->GetSampleRate())));
|
|
DEV("Length MS: %02d:%02d\n", (m_pSoundFile->GetLength() / 1000) / 60, (m_pSoundFile->GetLength() / 1000) % 60);
|
|
|
|
return;
|
|
}
|
|
}
|
|
|
|
CStream::~CStream()
|
|
{
|
|
Delete();
|
|
}
|
|
|
|
void CStream::Delete()
|
|
{
|
|
Stop();
|
|
ClearBuffers();
|
|
|
|
if ( m_pSoundFile )
|
|
{
|
|
delete m_pSoundFile;
|
|
m_pSoundFile = nil;
|
|
}
|
|
|
|
if ( m_pBuffer )
|
|
{
|
|
free(m_pBuffer);
|
|
m_pBuffer = nil;
|
|
}
|
|
}
|
|
|
|
bool CStream::HasSource()
|
|
{
|
|
return (m_pAlSources[0] != AL_NONE) && (m_pAlSources[1] != AL_NONE);
|
|
}
|
|
|
|
bool CStream::IsOpened()
|
|
{
|
|
return m_pSoundFile && m_pSoundFile->IsOpened();
|
|
}
|
|
|
|
bool CStream::IsPlaying()
|
|
{
|
|
if ( !HasSource() || !IsOpened() ) return false;
|
|
|
|
if ( !m_bPaused )
|
|
{
|
|
ALint sourceState[2];
|
|
alGetSourcei(m_pAlSources[0], AL_SOURCE_STATE, &sourceState[0]);
|
|
alGetSourcei(m_pAlSources[1], AL_SOURCE_STATE, &sourceState[1]);
|
|
if ( m_bActive || sourceState[0] == AL_PLAYING || sourceState[1] == AL_PLAYING)
|
|
return true;
|
|
}
|
|
|
|
return false;
|
|
}
|
|
|
|
void CStream::Pause()
|
|
{
|
|
if ( !HasSource() ) return;
|
|
ALint sourceState = AL_PAUSED;
|
|
alGetSourcei(m_pAlSources[0], AL_SOURCE_STATE, &sourceState);
|
|
if (sourceState != AL_PAUSED)
|
|
alSourcePause(m_pAlSources[0]);
|
|
alGetSourcei(m_pAlSources[1], AL_SOURCE_STATE, &sourceState);
|
|
if (sourceState != AL_PAUSED)
|
|
alSourcePause(m_pAlSources[1]);
|
|
}
|
|
|
|
void CStream::SetPause(bool bPause)
|
|
{
|
|
if ( !HasSource() ) return;
|
|
if ( bPause )
|
|
{
|
|
Pause();
|
|
m_bPaused = true;
|
|
}
|
|
else
|
|
{
|
|
if (m_bPaused)
|
|
SetPlay(true);
|
|
m_bPaused = false;
|
|
}
|
|
}
|
|
|
|
void CStream::SetPitch(float pitch)
|
|
{
|
|
if ( !HasSource() ) return;
|
|
alSourcef(m_pAlSources[0], AL_PITCH, pitch);
|
|
alSourcef(m_pAlSources[1], AL_PITCH, pitch);
|
|
}
|
|
|
|
void CStream::SetGain(float gain)
|
|
{
|
|
if ( !HasSource() ) return;
|
|
alSourcef(m_pAlSources[0], AL_GAIN, gain);
|
|
alSourcef(m_pAlSources[1], AL_GAIN, gain);
|
|
}
|
|
|
|
void CStream::SetPosition(int i, float x, float y, float z)
|
|
{
|
|
if ( !HasSource() ) return;
|
|
alSource3f(m_pAlSources[i], AL_POSITION, x, y, z);
|
|
}
|
|
|
|
void CStream::SetVolume(uint32 nVol)
|
|
{
|
|
m_nVolume = nVol;
|
|
SetGain(ALfloat(nVol) / MAX_VOLUME);
|
|
}
|
|
|
|
void CStream::SetPan(uint8 nPan)
|
|
{
|
|
m_nPan = clamp((int8)nPan - 63, 0, 63);
|
|
SetPosition(0, (m_nPan - 63) / 64.0f, 0.0f, Sqrt(1.0f - SQR((m_nPan - 63) / 64.0f)));
|
|
|
|
m_nPan = clamp((int8)nPan + 64, 64, 127);
|
|
SetPosition(1, (m_nPan - 63) / 64.0f, 0.0f, Sqrt(1.0f - SQR((m_nPan - 63) / 64.0f)));
|
|
|
|
m_nPan = nPan;
|
|
}
|
|
|
|
void CStream::SetPosMS(uint32 nPos)
|
|
{
|
|
if ( !IsOpened() ) return;
|
|
m_pSoundFile->Seek(nPos);
|
|
ClearBuffers();
|
|
}
|
|
|
|
uint32 CStream::GetPosMS()
|
|
{
|
|
if ( !HasSource() ) return 0;
|
|
if ( !IsOpened() ) return 0;
|
|
|
|
ALint offset;
|
|
//alGetSourcei(m_alSource, AL_SAMPLE_OFFSET, &offset);
|
|
alGetSourcei(m_pAlSources[0], AL_BYTE_OFFSET, &offset);
|
|
|
|
return m_pSoundFile->Tell()
|
|
- m_pSoundFile->samples2ms(m_pSoundFile->GetBufferSamples() * (NUM_STREAMBUFFERS/2-1)) / m_pSoundFile->GetChannels()
|
|
+ m_pSoundFile->samples2ms(offset/m_pSoundFile->GetSampleSize()) / m_pSoundFile->GetChannels();
|
|
}
|
|
|
|
uint32 CStream::GetLengthMS()
|
|
{
|
|
if ( !IsOpened() ) return 0;
|
|
return m_pSoundFile->GetLength();
|
|
}
|
|
|
|
bool CStream::FillBuffer(ALuint *alBuffer)
|
|
{
|
|
if ( !HasSource() )
|
|
return false;
|
|
if ( !IsOpened() )
|
|
return false;
|
|
if ( !(alBuffer[0] != AL_NONE && alIsBuffer(alBuffer[0])) )
|
|
return false;
|
|
if ( !(alBuffer[1] != AL_NONE && alIsBuffer(alBuffer[1])) )
|
|
return false;
|
|
|
|
uint32 size = m_pSoundFile->Decode(m_pBuffer);
|
|
if( size == 0 )
|
|
return false;
|
|
|
|
uint32 channelSize = size / m_pSoundFile->GetChannels();
|
|
|
|
alBufferData(alBuffer[0], AL_FORMAT_MONO16, m_pBuffer, channelSize, m_pSoundFile->GetSampleRate());
|
|
// TODO: use just one buffer if we play mono
|
|
if (m_pSoundFile->GetChannels() == 1)
|
|
alBufferData(alBuffer[1], AL_FORMAT_MONO16, m_pBuffer, channelSize, m_pSoundFile->GetSampleRate());
|
|
else
|
|
alBufferData(alBuffer[1], AL_FORMAT_MONO16, (uint8*)m_pBuffer + channelSize, channelSize, m_pSoundFile->GetSampleRate());
|
|
return true;
|
|
}
|
|
|
|
int32 CStream::FillBuffers()
|
|
{
|
|
int32 i = 0;
|
|
for ( i = 0; i < NUM_STREAMBUFFERS/2; i++ )
|
|
{
|
|
if ( !FillBuffer(&m_alBuffers[i*2]) )
|
|
break;
|
|
alSourceQueueBuffers(m_pAlSources[0], 1, &m_alBuffers[i*2]);
|
|
alSourceQueueBuffers(m_pAlSources[1], 1, &m_alBuffers[i*2+1]);
|
|
}
|
|
|
|
return i;
|
|
}
|
|
|
|
void CStream::ClearBuffers()
|
|
{
|
|
if ( !HasSource() ) return;
|
|
|
|
ALint buffersQueued[2];
|
|
alGetSourcei(m_pAlSources[0], AL_BUFFERS_QUEUED, &buffersQueued[0]);
|
|
alGetSourcei(m_pAlSources[1], AL_BUFFERS_QUEUED, &buffersQueued[1]);
|
|
|
|
ALuint value;
|
|
while (buffersQueued[0]--)
|
|
alSourceUnqueueBuffers(m_pAlSources[0], 1, &value);
|
|
while (buffersQueued[1]--)
|
|
alSourceUnqueueBuffers(m_pAlSources[1], 1, &value);
|
|
}
|
|
|
|
bool CStream::Setup()
|
|
{
|
|
if ( IsOpened() )
|
|
{
|
|
m_pSoundFile->Seek(0);
|
|
//SetPosition(0.0f, 0.0f, 0.0f);
|
|
SetPitch(1.0f);
|
|
//SetPan(m_nPan);
|
|
//SetVolume(100);
|
|
}
|
|
|
|
return IsOpened();
|
|
}
|
|
|
|
void CStream::SetPlay(bool state)
|
|
{
|
|
if ( !HasSource() ) return;
|
|
if ( state )
|
|
{
|
|
ALint sourceState = AL_PLAYING;
|
|
alGetSourcei(m_pAlSources[0], AL_SOURCE_STATE, &sourceState);
|
|
if (sourceState != AL_PLAYING )
|
|
alSourcePlay(m_pAlSources[0]);
|
|
|
|
sourceState = AL_PLAYING;
|
|
alGetSourcei(m_pAlSources[1], AL_SOURCE_STATE, &sourceState);
|
|
if (sourceState != AL_PLAYING)
|
|
alSourcePlay(m_pAlSources[1]);
|
|
|
|
m_bActive = true;
|
|
}
|
|
else
|
|
{
|
|
ALint sourceState = AL_STOPPED;
|
|
alGetSourcei(m_pAlSources[0], AL_SOURCE_STATE, &sourceState);
|
|
if (sourceState != AL_STOPPED )
|
|
alSourceStop(m_pAlSources[0]);
|
|
|
|
sourceState = AL_STOPPED;
|
|
alGetSourcei(m_pAlSources[1], AL_SOURCE_STATE, &sourceState);
|
|
if (sourceState != AL_STOPPED)
|
|
alSourceStop(m_pAlSources[1]);
|
|
|
|
m_bActive = false;
|
|
}
|
|
}
|
|
|
|
void CStream::Start()
|
|
{
|
|
if ( !HasSource() ) return;
|
|
if ( FillBuffers() != 0 )
|
|
SetPlay(true);
|
|
}
|
|
|
|
void CStream::Stop()
|
|
{
|
|
if ( !HasSource() ) return;
|
|
SetPlay(false);
|
|
}
|
|
|
|
void CStream::Update()
|
|
{
|
|
if ( !IsOpened() )
|
|
return;
|
|
|
|
if ( !HasSource() )
|
|
return;
|
|
|
|
if ( m_bReset )
|
|
return;
|
|
|
|
if ( !m_bPaused )
|
|
{
|
|
ALint sourceState[2];
|
|
ALint buffersProcessed[2] = { 0, 0 };
|
|
|
|
// 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);
|
|
|
|
if ( looping == AL_TRUE )
|
|
{
|
|
TRACE("stream set looping");
|
|
alSourcei(m_pAlSources[0], AL_LOOPING, AL_TRUE);
|
|
alSourcei(m_pAlSources[1], AL_LOOPING, AL_TRUE);
|
|
}
|
|
|
|
assert(buffersProcessed[0] == buffersProcessed[1]);
|
|
|
|
while( buffersProcessed[0]-- )
|
|
{
|
|
ALuint buffer[2];
|
|
|
|
alSourceUnqueueBuffers(m_pAlSources[0], 1, &buffer[0]);
|
|
alSourceUnqueueBuffers(m_pAlSources[1], 1, &buffer[1]);
|
|
|
|
if (m_bActive && FillBuffer(buffer))
|
|
{
|
|
alSourceQueueBuffers(m_pAlSources[0], 1, &buffer[0]);
|
|
alSourceQueueBuffers(m_pAlSources[1], 1, &buffer[1]);
|
|
}
|
|
}
|
|
|
|
if ( sourceState[0] != AL_PLAYING )
|
|
{
|
|
alGetSourcei(m_pAlSources[0], AL_BUFFERS_PROCESSED, &buffersProcessed[0]);
|
|
SetPlay(buffersProcessed[0]!=0);
|
|
}
|
|
}
|
|
}
|
|
|
|
void CStream::ProviderInit()
|
|
{
|
|
if ( m_bReset )
|
|
{
|
|
if ( Setup() )
|
|
{
|
|
SetPan(m_nPan);
|
|
SetVolume(m_nVolume);
|
|
SetPosMS(m_nPosBeforeReset);
|
|
if (m_bActive)
|
|
FillBuffers();
|
|
SetPlay(m_bActive);
|
|
if ( m_bPaused )
|
|
Pause();
|
|
}
|
|
|
|
m_bReset = false;
|
|
}
|
|
}
|
|
|
|
void CStream::ProviderTerm()
|
|
{
|
|
m_bReset = true;
|
|
m_nPosBeforeReset = GetPosMS();
|
|
|
|
ClearBuffers();
|
|
}
|
|
|
|
#endif
|