diff options
Diffstat (limited to '')
| -rw-r--r-- | src/audio/oal/stream.cpp | 811 | ||||
| -rw-r--r-- | src/audio/oal/stream.h | 10 |
2 files changed, 753 insertions, 68 deletions
diff --git a/src/audio/oal/stream.cpp b/src/audio/oal/stream.cpp index 90e90dd8..74ed86f4 100644 --- a/src/audio/oal/stream.cpp +++ b/src/audio/oal/stream.cpp @@ -4,22 +4,395 @@ #include "stream.h" #include "sampman.h" -#ifdef AUDIO_OPUS -#include <opusfile.h> -#else -#ifdef _WIN32 +#if defined _MSC_VER && !defined RE3_NO_AUTOLINK +#ifdef AUDIO_OAL_USE_SNDFILE #pragma comment( lib, "libsndfile-1.lib" ) +#endif +#ifdef AUDIO_OAL_USE_MPG123 #pragma comment( lib, "libmpg123-0.lib" ) #endif +#endif +#ifdef AUDIO_OAL_USE_SNDFILE #include <sndfile.h> +#endif +#ifdef AUDIO_OAL_USE_MPG123 #include <mpg123.h> #endif +#ifdef AUDIO_OAL_USE_OPUS +#include <opusfile.h> +#endif #ifndef _WIN32 #include "crossplatform.h" #endif -#ifndef AUDIO_OPUS +/* +As we ran onto an issue of having different volume levels for mono streams +and stereo streams we are now handling all the stereo panning ourselves. +Each stream now has two sources - one panned to the left and one to the right, +and uses two separate buffers to store data for each individual channel. +For that we also have to reshuffle all decoded PCM stereo data from LRLRLRLR to +LLLLRRRR (handled by CSortStereoBuffer). +*/ + +class CSortStereoBuffer +{ + uint16* PcmBuf; + size_t BufSize; +public: + CSortStereoBuffer() : PcmBuf(nil), BufSize(0) {} + ~CSortStereoBuffer() + { + if (PcmBuf) + free(PcmBuf); + } + + uint16* GetBuffer(size_t size) + { + if (size == 0) return nil; + if (!PcmBuf) + { + BufSize = size; + PcmBuf = (uint16*)malloc(BufSize); + } + else if (BufSize < size) + { + BufSize = size; + PcmBuf = (uint16*)realloc(PcmBuf, size); + } + return PcmBuf; + } + + void SortStereo(void* buf, size_t size) + { + uint16* InBuf = (uint16*)buf; + uint16* OutBuf = GetBuffer(size); + + if (!OutBuf) return; + + size_t rightStart = size / 4; + for (size_t i = 0; i < size / 4; i++) + { + OutBuf[i] = InBuf[i*2]; + OutBuf[i+rightStart] = InBuf[i*2+1]; + } + + memcpy(InBuf, OutBuf, size); + } + +}; + +CSortStereoBuffer SortStereoBuffer; + +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 *m_pFile; + bool m_bIsOpen; + + tFormatHeader m_FormatHeader; + + uint32 m_DataStartOffset; // TODO: 64 bit? + uint32 m_nSampleCount; + uint32 m_nSamplesPerBlock; + + // ADPCM things + uint8 *m_pAdpcmBuffer; + int16 **m_ppPcmBuffers; + CImaADPCMDecoder *m_pAdpcmDecoders; + + void Close() + { + if (m_pFile) { + fclose(m_pFile); + m_pFile = nil; + } + delete[] m_pAdpcmBuffer; + delete[] m_ppPcmBuffers; + delete[] m_pAdpcmDecoders; + } + + uint32 GetCurrentSample() const + { + // TODO: 64 bit? + uint32 FilePos = ftell(m_pFile); + if (FilePos <= m_DataStartOffset) + return 0; + return (FilePos - m_DataStartOffset) / m_FormatHeader.BlockAlign * m_nSamplesPerBlock; + } + +public: + 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) + { + m_pFile = fopen(path, "rb"); + if (!m_pFile) return; + +#define CLOSE_ON_ERROR(op)\ + if (op) { \ + Close(); \ + return; \ + } + + tDataHeader DataHeader; + + CLOSE_ON_ERROR(fread(&DataHeader, sizeof(DataHeader), 1, m_pFile) == 0); + CLOSE_ON_ERROR(DataHeader.ID != 'FFIR'); + + // TODO? validate filesizes + + int WAVE; + CLOSE_ON_ERROR(fread(&WAVE, 4, 1, m_pFile) == 0); + CLOSE_ON_ERROR(WAVE != 'EVAW') + CLOSE_ON_ERROR(fread(&DataHeader, sizeof(DataHeader), 1, m_pFile) == 0); + CLOSE_ON_ERROR(DataHeader.ID != ' tmf'); + + CLOSE_ON_ERROR(fread(&m_FormatHeader, Min(DataHeader.Size, sizeof(tFormatHeader)), 1, m_pFile) == 0); + CLOSE_ON_ERROR(DataHeader.Size > sizeof(tFormatHeader)); + + switch (m_FormatHeader.AudioFormat) + { + case WAVEFMT_XBOX_ADPCM: + m_FormatHeader.AudioFormat = WAVEFMT_IMA_ADPCM; + case WAVEFMT_IMA_ADPCM: + m_nSamplesPerBlock = (m_FormatHeader.BlockAlign / m_FormatHeader.NumChannels - 4) * 2 + 1; + m_pAdpcmBuffer = new uint8[m_FormatHeader.BlockAlign]; + m_ppPcmBuffers = new int16*[m_FormatHeader.NumChannels]; + m_pAdpcmDecoders = new CImaADPCMDecoder[m_FormatHeader.NumChannels]; + break; + case WAVEFMT_PCM: + m_nSamplesPerBlock = 1; + if (m_FormatHeader.BitsPerSample != 16) + { + debug("Unsupported PCM (%d bits), only signed 16-bit is supported (%s)\n", m_FormatHeader.BitsPerSample, path); + Close(); + return; + } + break; + default: + debug("Unsupported wav format 0x%x (%s)\n", m_FormatHeader.AudioFormat, path); + Close(); + return; + } + + while (true) { + CLOSE_ON_ERROR(fread(&DataHeader, sizeof(DataHeader), 1, m_pFile) == 0); + if (DataHeader.ID == 'atad') + break; + fseek(m_pFile, DataHeader.Size, SEEK_CUR); + // TODO? validate data size + // maybe check if there no extreme custom headers that might break this + } + + m_DataStartOffset = ftell(m_pFile); + m_nSampleCount = DataHeader.Size / m_FormatHeader.BlockAlign * m_nSamplesPerBlock; + + m_bIsOpen = true; +#undef CLOSE_ON_ERROR + } + + ~CWavFile() + { + Close(); + } + + bool IsOpened() + { + return m_bIsOpen; + } + + uint32 GetSampleSize() + { + return sizeof(uint16); + } + + uint32 GetSampleCount() + { + return m_nSampleCount; + } + + uint32 GetSampleRate() + { + return m_FormatHeader.SampleRate; + } + + uint32 GetChannels() + { + return m_FormatHeader.NumChannels; + } + + void Seek(uint32 milliseconds) + { + if (!IsOpened()) return; + fseek(m_pFile, m_DataStartOffset + ms2samples(milliseconds) / m_nSamplesPerBlock * m_FormatHeader.BlockAlign, SEEK_SET); + } + + uint32 Tell() + { + if (!IsOpened()) return 0; + return samples2ms(GetCurrentSample()); + } + +#define SAMPLES_IN_LINE (8) + + uint32 Decode(void* buffer) + { + if (!IsOpened()) return 0; + + if (m_FormatHeader.AudioFormat == WAVEFMT_PCM) + { + // just read the file and sort the samples + uint32 size = fread(buffer, 1, GetBufferSize(), m_pFile); + if (m_FormatHeader.NumChannels == 2) + SortStereoBuffer.SortStereo(buffer, size); + return size; + } + else if (m_FormatHeader.AudioFormat == WAVEFMT_IMA_ADPCM) + { + // trim the buffer size if we're at the end of our file + uint32 nMaxSamples = GetBufferSamples() / m_FormatHeader.NumChannels; + uint32 nSamplesLeft = m_nSampleCount - GetCurrentSample(); + nMaxSamples = Min(nMaxSamples, nSamplesLeft); + + // align sample count to our block + nMaxSamples = nMaxSamples / m_nSamplesPerBlock * m_nSamplesPerBlock; + + // count the size of output buffer + uint32 OutBufSizePerChannel = nMaxSamples * GetSampleSize(); + uint32 OutBufSize = OutBufSizePerChannel * m_FormatHeader.NumChannels; + + // calculate the pointers to individual channel buffers + for (uint32 i = 0; i < m_FormatHeader.NumChannels; i++) + m_ppPcmBuffers[i] = (int16*)((int8*)buffer + OutBufSizePerChannel * i); + + uint32 samplesRead = 0; + while (samplesRead < nMaxSamples) + { + // read the file + uint8 *pAdpcmBuf = m_pAdpcmBuffer; + if (fread(m_pAdpcmBuffer, 1, m_FormatHeader.BlockAlign, m_pFile) == 0) + return 0; + + // get the first sample in adpcm block and initialise the decoder(s) + for (uint32 i = 0; i < m_FormatHeader.NumChannels; i++) + { + int16 Sample = *(int16*)pAdpcmBuf; + pAdpcmBuf += sizeof(int16); + int16 Step = *(int16*)pAdpcmBuf; + pAdpcmBuf += sizeof(int16); + m_pAdpcmDecoders[i].Init(Sample, Step); + *(m_ppPcmBuffers[i]) = Sample; + m_ppPcmBuffers[i]++; + } + samplesRead++; + + // decode the rest of the block + for (uint32 s = 1; s < m_nSamplesPerBlock; s += SAMPLES_IN_LINE) + { + for (uint32 i = 0; i < m_FormatHeader.NumChannels; i++) + { + m_pAdpcmDecoders[i].Decode(pAdpcmBuf, m_ppPcmBuffers[i], SAMPLES_IN_LINE / 2); + pAdpcmBuf += SAMPLES_IN_LINE / 2; + m_ppPcmBuffers[i] += SAMPLES_IN_LINE; + } + samplesRead += SAMPLES_IN_LINE; + } + } + return OutBufSize; + } + return 0; + } +}; + +#ifdef AUDIO_OAL_USE_SNDFILE class CSndFile : public IDecoder { SNDFILE *m_pfSound; @@ -81,9 +454,18 @@ public: uint32 Decode(void *buffer) { if ( !IsOpened() ) return 0; - return sf_read_short(m_pfSound, (short *)buffer, GetBufferSamples()) * GetSampleSize(); + + size_t size = sf_read_short(m_pfSound, (short*)buffer, GetBufferSamples()) * GetSampleSize(); + if (GetChannels()==2) + SortStereoBuffer.SortStereo(buffer, size); + return size; } }; +#endif + +#ifdef AUDIO_OAL_USE_MPG123 +// fuzzy seek eliminates stutter when playing ADF but spams errors a lot (nothing breaks though) +#define MP3_USE_FUZZY_SEEK class CMP3File : public IDecoder { @@ -101,6 +483,9 @@ public: 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; @@ -176,10 +561,251 @@ public: 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; } }; -#else + +#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; @@ -267,6 +893,9 @@ public: if (size < 0) return 0; + if (GetChannels() == 2) + SortStereoBuffer.SortStereo(buffer, size * m_nChannels * GetSampleSize()); + return size * m_nChannels * GetSampleSize(); } }; @@ -274,20 +903,20 @@ public: void CStream::Initialise() { -#ifndef AUDIO_OPUS +#ifdef AUDIO_OAL_USE_MPG123 mpg123_init(); #endif } void CStream::Terminate() { -#ifndef AUDIO_OPUS +#ifdef AUDIO_OAL_USE_MPG123 mpg123_exit(); #endif } -CStream::CStream(char *filename, ALuint &source, ALuint (&buffers)[NUM_STREAMBUFFERS]) : - m_alSource(source), +CStream::CStream(char *filename, ALuint *sources, ALuint (&buffers)[NUM_STREAMBUFFERS], uint32 overrideSampleRate) : + m_pAlSources(sources), m_alBuffers(buffers), m_pBuffer(nil), m_bPaused(false), @@ -314,13 +943,20 @@ CStream::CStream(char *filename, ALuint &source, ALuint (&buffers)[NUM_STREAMBUF DEV("Stream %s\n", m_aFilename); -#ifndef AUDIO_OPUS - 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")) + if (!strcasecmp(&m_aFilename[strlen(m_aFilename) - strlen(".wav")], ".wav")) +#ifdef AUDIO_OAL_USE_SNDFILE m_pSoundFile = new CSndFile(m_aFilename); #else - if (!strcasecmp(&m_aFilename[strlen(m_aFilename) - strlen(".opus")], ".opus")) + 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); +#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 @@ -368,7 +1004,7 @@ void CStream::Delete() bool CStream::HasSource() { - return m_alSource != AL_NONE; + return (m_pAlSources[0] != AL_NONE) && (m_pAlSources[1] != AL_NONE); } bool CStream::IsOpened() @@ -382,9 +1018,10 @@ bool CStream::IsPlaying() if ( !m_bPaused ) { - ALint sourceState; - alGetSourcei(m_alSource, AL_SOURCE_STATE, &sourceState); - if ( m_bActive || sourceState == AL_PLAYING ) + 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; } @@ -395,9 +1032,12 @@ void CStream::Pause() { if ( !HasSource() ) return; ALint sourceState = AL_PAUSED; - alGetSourcei(m_alSource, AL_SOURCE_STATE, &sourceState); - if (sourceState != AL_PAUSED ) - alSourcePause(m_alSource); + 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) @@ -419,19 +1059,21 @@ void CStream::SetPause(bool bPause) void CStream::SetPitch(float pitch) { if ( !HasSource() ) return; - alSourcef(m_alSource, AL_PITCH, pitch); + alSourcef(m_pAlSources[0], AL_PITCH, pitch); + alSourcef(m_pAlSources[1], AL_PITCH, pitch); } void CStream::SetGain(float gain) { if ( !HasSource() ) return; - alSourcef(m_alSource, AL_GAIN, gain); + alSourcef(m_pAlSources[0], AL_GAIN, gain); + alSourcef(m_pAlSources[1], AL_GAIN, gain); } -void CStream::SetPosition(float x, float y, float z) +void CStream::SetPosition(int i, float x, float y, float z) { if ( !HasSource() ) return; - alSource3f(m_alSource, AL_POSITION, x, y, z); + alSource3f(m_pAlSources[i], AL_POSITION, x, y, z); } void CStream::SetVolume(uint32 nVol) @@ -442,8 +1084,13 @@ void CStream::SetVolume(uint32 nVol) 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; - SetPosition((nPan - 63)/64.0f, 0.0f, Sqrt(1.0f-SQR((nPan-63)/64.0f))); } void CStream::SetPosMS(uint32 nPos) @@ -460,10 +1107,10 @@ uint32 CStream::GetPosMS() ALint offset; //alGetSourcei(m_alSource, AL_SAMPLE_OFFSET, &offset); - alGetSourcei(m_alSource, AL_BYTE_OFFSET, &offset); + alGetSourcei(m_pAlSources[0], AL_BYTE_OFFSET, &offset); return m_pSoundFile->Tell() - - m_pSoundFile->samples2ms(m_pSoundFile->GetBufferSamples() * (NUM_STREAMBUFFERS-1)) / m_pSoundFile->GetChannels() + - m_pSoundFile->samples2ms(m_pSoundFile->GetBufferSamples() * (NUM_STREAMBUFFERS/2-1)) / m_pSoundFile->GetChannels() + m_pSoundFile->samples2ms(offset/m_pSoundFile->GetSampleSize()) / m_pSoundFile->GetChannels(); } @@ -473,33 +1120,41 @@ uint32 CStream::GetLengthMS() return m_pSoundFile->GetLength(); } -bool CStream::FillBuffer(ALuint alBuffer) +bool CStream::FillBuffer(ALuint *alBuffer) { if ( !HasSource() ) return false; if ( !IsOpened() ) return false; - if ( !(alBuffer != AL_NONE && alIsBuffer(alBuffer)) ) + 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, m_pSoundFile->GetChannels() == 1 ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16, - m_pBuffer, size, m_pSoundFile->GetSampleRate()); - + 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; i++ ) + for ( i = 0; i < NUM_STREAMBUFFERS/2; i++ ) { - if ( !FillBuffer(m_alBuffers[i]) ) + if ( !FillBuffer(&m_alBuffers[i*2]) ) break; - alSourceQueueBuffers(m_alSource, 1, &m_alBuffers[i]); + alSourceQueueBuffers(m_pAlSources[0], 1, &m_alBuffers[i*2]); + alSourceQueueBuffers(m_pAlSources[1], 1, &m_alBuffers[i*2+1]); } return i; @@ -508,13 +1163,16 @@ int32 CStream::FillBuffers() void CStream::ClearBuffers() { if ( !HasSource() ) return; - - ALint buffersQueued; - alGetSourcei(m_alSource, AL_BUFFERS_QUEUED, &buffersQueued); + + 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--) - alSourceUnqueueBuffers(m_alSource, 1, &value); + while (buffersQueued[0]--) + alSourceUnqueueBuffers(m_pAlSources[0], 1, &value); + while (buffersQueued[1]--) + alSourceUnqueueBuffers(m_pAlSources[1], 1, &value); } bool CStream::Setup() @@ -522,7 +1180,6 @@ bool CStream::Setup() if ( IsOpened() ) { m_pSoundFile->Seek(0); - alSourcei(m_alSource, AL_SOURCE_RELATIVE, AL_TRUE); //SetPosition(0.0f, 0.0f, 0.0f); SetPitch(1.0f); //SetPan(m_nPan); @@ -538,17 +1195,29 @@ void CStream::SetPlay(bool state) if ( state ) { ALint sourceState = AL_PLAYING; - alGetSourcei(m_alSource, AL_SOURCE_STATE, &sourceState); + alGetSourcei(m_pAlSources[0], AL_SOURCE_STATE, &sourceState); if (sourceState != AL_PLAYING ) - alSourcePlay(m_alSource); + 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_alSource, AL_SOURCE_STATE, &sourceState); - if (sourceState != AL_STOPPED ) - alSourceStop(m_alSource); + 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; } } @@ -579,35 +1248,51 @@ void CStream::Update() if ( !m_bPaused ) { - ALint sourceState; - ALint buffersProcessed = 0; + ALint sourceState[2]; + ALint buffersProcessed[2] = { 0, 0 }; - alGetSourcei(m_alSource, AL_SOURCE_STATE, &sourceState); - alGetSourcei(m_alSource, AL_BUFFERS_PROCESSED, &buffersProcessed); + // 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_alSource, AL_LOOPING, &looping); + alGetSourcei(m_pAlSources[0], AL_LOOPING, &looping); if ( looping == AL_TRUE ) { TRACE("stream set looping"); - alSourcei(m_alSource, AL_LOOPING, AL_TRUE); + alSourcei(m_pAlSources[0], AL_LOOPING, AL_TRUE); + alSourcei(m_pAlSources[1], AL_LOOPING, AL_TRUE); } + + assert(buffersProcessed[0] == buffersProcessed[1]); - while( buffersProcessed-- ) + while( buffersProcessed[0]-- ) { - ALuint buffer; + ALuint buffer[2]; - alSourceUnqueueBuffers(m_alSource, 1, &buffer); + alSourceUnqueueBuffers(m_pAlSources[0], 1, &buffer[0]); + alSourceUnqueueBuffers(m_pAlSources[1], 1, &buffer[1]); - if ( m_bActive && FillBuffer(buffer) ) - alSourceQueueBuffers(m_alSource, 1, &buffer); + if (m_bActive && FillBuffer(buffer)) + { + alSourceQueueBuffers(m_pAlSources[0], 1, &buffer[0]); + alSourceQueueBuffers(m_pAlSources[1], 1, &buffer[1]); + } } - if ( sourceState != AL_PLAYING ) + if ( sourceState[0] != AL_PLAYING ) { - alGetSourcei(m_alSource, AL_BUFFERS_PROCESSED, &buffersProcessed); - SetPlay(buffersProcessed!=0); + alGetSourcei(m_pAlSources[0], AL_BUFFERS_PROCESSED, &buffersProcessed[0]); + SetPlay(buffersProcessed[0]!=0); } } } diff --git a/src/audio/oal/stream.h b/src/audio/oal/stream.h index 2476abcc..bcbc5e54 100644 --- a/src/audio/oal/stream.h +++ b/src/audio/oal/stream.h @@ -3,7 +3,7 @@ #ifdef AUDIO_OAL #include <AL/al.h> -#define NUM_STREAMBUFFERS 4 +#define NUM_STREAMBUFFERS 8 class IDecoder { @@ -57,7 +57,7 @@ public: class CStream { char m_aFilename[128]; - ALuint &m_alSource; + ALuint *m_pAlSources; ALuint (&m_alBuffers)[NUM_STREAMBUFFERS]; bool m_bPaused; @@ -73,20 +73,20 @@ class CStream IDecoder *m_pSoundFile; bool HasSource(); - void SetPosition(float x, float y, float z); + void SetPosition(int i, float x, float y, float z); void SetPitch(float pitch); void SetGain(float gain); void Pause(); void SetPlay(bool state); - bool FillBuffer(ALuint alBuffer); + bool FillBuffer(ALuint *alBuffer); int32 FillBuffers(); void ClearBuffers(); public: static void Initialise(); static void Terminate(); - CStream(char *filename, ALuint &source, ALuint (&buffers)[NUM_STREAMBUFFERS]); + CStream(char *filename, ALuint *sources, ALuint (&buffers)[NUM_STREAMBUFFERS], uint32 overrideSampleRate = 32000); ~CStream(); void Delete(); |