renamed eof -> eos (end of stream); hid MP3/Vorbis stream classes completly (by providing factory methods); new readBuffer method for AudioInputStream for improved speed of the mixer; new MusicStream class (subclassed for MP3/Vorbis sound) which offers a getRate method; some other tweaks
svn-id: r9467
This commit is contained in:
Max Horn
parent
679e818b0b
commit
d280258e09
4 changed files with 214 additions and 103 deletions
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@ -47,6 +47,17 @@ protected:
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const byte *_loopPtr;
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const byte *_loopEnd;
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inline int16 readIntern() {
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//assert(_ptr < _end);
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int16 val = readSample<is16Bit, isUnsigned>(_ptr);
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_ptr += (is16Bit ? 2 : 1);
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if (_loopPtr && _ptr == _end) {
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_ptr = _loopPtr;
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_end = _loopEnd;
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}
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return val;
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}
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inline bool eosIntern() const { return _ptr >= _end; };
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public:
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LinearMemoryStream(const byte *ptr, uint len, uint loopOffset, uint loopLen)
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: _ptr(ptr), _end(ptr+len), _loopPtr(0), _loopEnd(0) {
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@ -57,22 +68,25 @@ public:
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if (stereo) // Stereo requires even sized data
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assert(len % 2 == 0);
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}
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int16 read() {
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//assert(_ptr < _end);
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int16 val = readSample<is16Bit, isUnsigned>(_ptr);
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_ptr += (is16Bit ? 2 : 1);
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if (_loopPtr && _ptr == _end) {
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_ptr = _loopPtr;
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_end = _loopEnd;
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}
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return val;
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}
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bool eof() const {
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return _ptr >= _end;
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}
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bool isStereo() const {
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return stereo;
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int readBuffer(int16 *buffer, int numSamples) {
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int samples = 0;
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do {
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const int len = MIN(numSamples, (_end - _ptr) / (is16Bit ? 2 : 1));
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for (; samples < len; samples++) {
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*buffer++ = readSample<is16Bit, isUnsigned>(_ptr);
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_ptr += (is16Bit ? 2 : 1);
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}
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if (_loopPtr && _ptr == _end) {
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_ptr = _loopPtr;
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_end = _loopEnd;
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}
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} while (samples < numSamples && !eosIntern());
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return samples;
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}
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int16 read() { return readIntern(); }
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bool eos() const { return eosIntern(); }
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bool isStereo() const { return stereo; }
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};
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@ -90,14 +104,16 @@ protected:
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byte *_pos;
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byte *_end;
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inline int16 readIntern();
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inline bool eosIntern() const { return _end == _pos; };
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public:
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WrappedMemoryStream(uint bufferSize);
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~WrappedMemoryStream() { free(_bufferStart); }
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int16 read();
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bool eof() const;
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bool isStereo() const {
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return stereo;
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}
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~WrappedMemoryStream() { free(_bufferStart); }
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int readBuffer(int16 *buffer, int numSamples);
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int16 read() { return readIntern(); }
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bool eos() const { return eosIntern(); }
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bool isStereo() const { return stereo; }
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void append(const byte *data, uint32 len);
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};
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@ -113,7 +129,7 @@ WrappedMemoryStream<stereo, is16Bit, isUnsigned>::WrappedMemoryStream(uint buffe
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}
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template<bool stereo, bool is16Bit, bool isUnsigned>
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int16 WrappedMemoryStream<stereo, is16Bit, isUnsigned>::read() {
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inline int16 WrappedMemoryStream<stereo, is16Bit, isUnsigned>::readIntern() {
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//assert(_pos != _end);
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int16 val = readSample<is16Bit, isUnsigned>(_pos);
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_pos += (is16Bit ? 2 : 1);
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@ -126,8 +142,12 @@ int16 WrappedMemoryStream<stereo, is16Bit, isUnsigned>::read() {
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}
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template<bool stereo, bool is16Bit, bool isUnsigned>
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bool WrappedMemoryStream<stereo, is16Bit, isUnsigned>::eof() const {
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return _end == _pos;
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int WrappedMemoryStream<stereo, is16Bit, isUnsigned>::readBuffer(int16 *buffer, int numSamples) {
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int samples;
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for (samples = 0; samples < numSamples && !eosIntern(); samples++) {
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*buffer++ = readIntern();
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}
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return samples;
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}
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template<bool stereo, bool is16Bit, bool isUnsigned>
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@ -160,6 +180,37 @@ void WrappedMemoryStream<stereo, is16Bit, isUnsigned>::append(const byte *data,
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#ifdef USE_MAD
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class MP3InputStream : public MusicStream {
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struct mad_stream _stream;
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struct mad_frame _frame;
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struct mad_synth _synth;
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mad_timer_t _duration;
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uint32 _posInFrame;
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uint32 _bufferSize;
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int _size;
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bool _isStereo;
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int _curChannel;
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File *_file;
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byte *_ptr;
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int _rate;
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bool _initialized;
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bool init();
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void refill();
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inline int16 readIntern();
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inline bool eosIntern() const;
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public:
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MP3InputStream(File *file, mad_timer_t duration, uint size = 0);
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~MP3InputStream();
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int readBuffer(int16 *buffer, int numSamples);
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int16 read() { return readIntern(); }
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bool eos() const { return eosIntern(); }
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bool isStereo() const { return _isStereo; }
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int getRate() const { return _rate; }
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};
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/**
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* Playback the MP3 data in the given file for the specified duration.
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@ -299,7 +350,7 @@ void MP3InputStream::refill() {
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_posInFrame = 0;
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}
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bool MP3InputStream::eof() const {
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inline bool MP3InputStream::eosIntern() const {
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return (_size < 0 || _posInFrame >= _synth.pcm.length);
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}
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@ -317,7 +368,7 @@ static inline int scale_sample(mad_fixed_t sample) {
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return sample >> (MAD_F_FRACBITS + 1 - 16);
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}
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int16 MP3InputStream::read() {
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inline int16 MP3InputStream::readIntern() {
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if (_size < 0 || _posInFrame >= _synth.pcm.length) { // EOF
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return 0;
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}
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@ -343,6 +394,18 @@ int16 MP3InputStream::read() {
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return sample;
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}
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int MP3InputStream::readBuffer(int16 *buffer, int numSamples) {
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int samples;
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for (samples = 0; samples < numSamples && !eosIntern(); samples++) {
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*buffer++ = readIntern();
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}
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return samples;
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}
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MusicStream *makeMP3Stream(File *file, mad_timer_t duration, uint size) {
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return new MP3InputStream(file, duration, size);
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}
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#endif
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@ -353,6 +416,29 @@ int16 MP3InputStream::read() {
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#ifdef USE_VORBIS
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class VorbisInputStream : public MusicStream {
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OggVorbis_File *_ov_file;
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int _end_pos;
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bool _eofFlag;
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int _numChannels;
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int16 _buffer[4096];
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int16 *_pos;
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void refill();
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inline int16 readIntern();
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inline bool eosIntern() const;
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public:
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VorbisInputStream(OggVorbis_File *file, int duration);
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int readBuffer(int16 *buffer, int numSamples);
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int16 read() { return readIntern(); }
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bool eos() const { return eosIntern(); }
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bool isStereo() const { return _numChannels >= 2; }
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int getRate() const { return ov_info(_ov_file, -1)->rate; }
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};
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#ifdef CHUNKSIZE
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#define VORBIS_TREMOR
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#endif
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@ -371,14 +457,14 @@ VorbisInputStream::VorbisInputStream(OggVorbis_File *file, int duration)
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_eofFlag = false;
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}
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int16 VorbisInputStream::read() {
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inline int16 VorbisInputStream::readIntern() {
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if (_pos >= _buffer + ARRAYSIZE(_buffer)) {
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refill();
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}
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return *_pos++;
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}
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bool VorbisInputStream::eof() const {
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inline bool VorbisInputStream::eosIntern() const {
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if (_eofFlag)
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return true;
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if (_pos < _buffer + ARRAYSIZE(_buffer))
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@ -386,6 +472,14 @@ bool VorbisInputStream::eof() const {
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return (_end_pos <= ov_pcm_tell(_ov_file));
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}
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int VorbisInputStream::readBuffer(int16 *buffer, int numSamples) {
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int samples;
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for (samples = 0; samples < numSamples && !eosIntern(); samples++) {
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*buffer++ = readIntern();
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}
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return samples;
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}
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void VorbisInputStream::refill() {
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// Read the samples
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uint len_left = sizeof(_buffer);
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@ -426,6 +520,10 @@ void VorbisInputStream::refill() {
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_pos = _buffer;
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}
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MusicStream *makeVorbisStream(OggVorbis_File *file, int duration) {
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return new VorbisInputStream(file, duration);
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}
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#endif
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@ -24,6 +24,7 @@
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#include "stdafx.h"
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#include "common/scummsys.h"
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#include "common/util.h"
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#ifdef USE_MAD
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#include <mad.h>
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#endif
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@ -47,10 +48,34 @@ class AudioInputStream {
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public:
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virtual ~AudioInputStream() {}
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/**
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* Fill the given buffer with up to numSamples samples.
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* Returns the actual number of samples read, or -1 if
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* a critical error occured (note: you *must* check if
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* this value is less than what you requested, this can
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* happend when the stream is fully used up).
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* For stereo stream, buffer will be filled with interleaved
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* left and right channel samples.
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*
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* For maximum efficency, subclasses should always override
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* the default implementation!
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*/
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virtual int readBuffer(int16 *buffer, int numSamples) {
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int samples;
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for (samples = 0; samples < numSamples && !eos(); samples++) {
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*buffer++ = read();
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}
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return samples;
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}
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/** Read a singel (16 bit signed) sample from the stream. */
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virtual int16 read() = 0;
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//virtual int size() const = 0;
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/** Is this a stereo stream? */
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virtual bool isStereo() const = 0;
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virtual bool eof() const = 0;
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/* End of stream reached? */
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virtual bool eos() const = 0;
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virtual int getRate() const { return -1; }
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};
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@ -65,64 +90,34 @@ protected:
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int _len;
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public:
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ZeroInputStream(uint len) : _len(len) { }
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int readBuffer(int16 *buffer, int numSamples) {
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int samples = MIN(_len, numSamples);
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memset(buffer, 0, samples * 2);
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_len -= samples;
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return samples;
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}
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int16 read() { assert(_len > 0); _len--; return 0; }
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int size() const { return _len; }
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bool isStereo() const { return false; }
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bool eof() const { return _len <= 0; }
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bool eos() const { return _len <= 0; }
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};
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#ifdef USE_MAD
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class MP3InputStream : public AudioInputStream {
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struct mad_stream _stream;
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struct mad_frame _frame;
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struct mad_synth _synth;
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mad_timer_t _duration;
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uint32 _posInFrame;
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uint32 _bufferSize;
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int _size;
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bool _isStereo;
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int _curChannel;
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File *_file;
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byte *_ptr;
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int _rate;
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bool _initialized;
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bool init();
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void refill();
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class MusicStream : public AudioInputStream {
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public:
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MP3InputStream(File *file, mad_timer_t duration, uint size = 0);
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~MP3InputStream();
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int16 read();
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bool eof() const;
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bool isStereo() const { return _isStereo; }
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int getRate() const { return _rate; }
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virtual int getRate() const = 0;
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};
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#endif
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#ifdef USE_VORBIS
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class VorbisInputStream : public AudioInputStream {
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OggVorbis_File *_ov_file;
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int _end_pos;
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bool _eofFlag;
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int _numChannels;
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int16 _buffer[4096];
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int16 *_pos;
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void refill();
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public:
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VorbisInputStream(OggVorbis_File *file, int duration);
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int16 read();
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bool eof() const;
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bool isStereo() const { return _numChannels >= 2; }
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};
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#endif
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AudioInputStream *makeLinearInputStream(byte _flags, const byte *ptr, uint32 len, uint loopOffset, uint loopLen);
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WrappedAudioInputStream *makeWrappedInputStream(byte _flags, uint32 len);
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#ifdef USE_MAD
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MusicStream *makeMP3Stream(File *file, mad_timer_t duration, uint size = 0);
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#endif
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#ifdef USE_VORBIS
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MusicStream *makeVorbisStream(OggVorbis_File *file, int duration);
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#endif
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#endif
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@ -709,7 +709,7 @@ void ChannelRaw::mix(int16 *data, uint len) {
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assert(_input);
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assert(_converter);
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if (_input->eof()) {
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if (_input->eos()) {
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// TODO: call drain method
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destroy();
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return;
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@ -820,7 +820,7 @@ void ChannelStream::mix(int16 *data, uint len) {
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assert(_input);
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assert(_converter);
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if (_input->eof()) {
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if (_input->eos()) {
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// TODO: call drain method
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// Normally, the stream stays around even if all its data is used up.
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@ -929,7 +929,7 @@ static inline int scale_sample(mad_fixed_t sample) {
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ChannelMP3::ChannelMP3(SoundMixer *mixer, PlayingSoundHandle *handle, File *file, uint size)
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: Channel(mixer, handle) {
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// Create the input stream
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_input = new MP3InputStream(file, mad_timer_zero, size);
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_input = makeMP3Stream(file, mad_timer_zero, size);
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// Get a rate converter instance
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//printf("ChannelMP3: inrate %d, outrate %d, stereo %d\n", _input->getRate(), mixer->getOutputRate(), _input->isStereo());
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@ -951,7 +951,7 @@ void ChannelMP3::mix(int16 *data, uint len) {
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assert(_input);
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assert(_converter);
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if (_input->eof()) {
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if (_input->eos()) {
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// TODO: call drain method
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destroy();
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return;
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@ -1013,7 +1013,7 @@ void ChannelMP3::mix(int16 *data, uint len) {
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ChannelMP3CDMusic::ChannelMP3CDMusic(SoundMixer *mixer, PlayingSoundHandle *handle, File *file, mad_timer_t duration)
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: Channel(mixer, handle) {
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// Create the input stream
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_input = new MP3InputStream(file, duration, 0);
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_input = makeMP3Stream(file, duration, 0);
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// Get a rate converter instance
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//printf("ChannelMP3CDMusic: inrate %d, outrate %d, stereo %d\n", _input->getRate(), mixer->getOutputRate(), _input->isStereo());
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@ -1034,7 +1034,7 @@ void ChannelMP3CDMusic::mix(int16 *data, uint len) {
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assert(_input);
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assert(_converter);
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if (_input->eof()) {
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if (_input->eos()) {
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// TODO: call drain method
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destroy();
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return;
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@ -1158,15 +1158,11 @@ void ChannelMP3CDMusic::mix(int16 *data, uint len) {
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ChannelVorbis::ChannelVorbis(SoundMixer *mixer, PlayingSoundHandle *handle, OggVorbis_File *ov_file, int duration, bool is_cd_track)
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: Channel(mixer, handle) {
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#ifdef SOX_HACK
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vorbis_info *vi;
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// Create the input stream
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_input = new VorbisInputStream(ov_file, duration);
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_input = makeVorbisStream(ov_file, duration);
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// Get a rate converter instance
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vi = ov_info(ov_file, -1);
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assert(vi->channels == 1 || vi->channels == 2);
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_converter = makeRateConverter(vi->rate, mixer->getOutputRate(), _input->isStereo());
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_converter = makeRateConverter(_input->getRate(), mixer->getOutputRate(), _input->isStereo());
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#else
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_ov_file = ov_file;
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@ -1187,7 +1183,7 @@ void ChannelVorbis::mix(int16 *data, uint len) {
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assert(_input);
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assert(_converter);
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if (_input->eof()) {
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if (_input->eos()) {
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// TODO: call drain method
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destroy();
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return;
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@ -30,9 +30,20 @@
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#include "stdafx.h"
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#include "sound/rate.h"
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/**
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* The precision of the fractional computations used by the rate converter.
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* Normally you should never have to modify this value.
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*/
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#define FRAC_BITS 16
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/**
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* The size of the intermediate input cache. Bigger values may increase
|
||||
* performance, but only until some point (depends largely on cache size,
|
||||
* target processor and various other factors), at which it will decrease
|
||||
* again.
|
||||
*/
|
||||
#define INTERMEDIATE_BUFFER_SIZE 512
|
||||
|
||||
|
||||
/**
|
||||
* Audio rate converter based on simple linear Interpolation.
|
||||
|
|
@ -48,7 +59,9 @@
|
|||
template<bool stereo, bool reverseStereo>
|
||||
class LinearRateConverter : public RateConverter {
|
||||
protected:
|
||||
bool _reverseStereo;
|
||||
st_sample_t inBuf[INTERMEDIATE_BUFFER_SIZE];
|
||||
const st_sample_t *inPtr;
|
||||
int inLen;
|
||||
|
||||
/** fractional position of the output stream in input stream unit */
|
||||
unsigned long opos, opos_frac;
|
||||
|
|
@ -101,6 +114,8 @@ LinearRateConverter<stereo, reverseStereo>::LinearRateConverter(st_rate_t inrate
|
|||
|
||||
ilast[0] = ilast[1] = 0;
|
||||
icur[0] = icur[1] = 0;
|
||||
|
||||
inLen = 0;
|
||||
}
|
||||
|
||||
/*
|
||||
|
|
@ -112,6 +127,9 @@ int LinearRateConverter<stereo, reverseStereo>::flow(AudioInputStream &input, st
|
|||
{
|
||||
st_sample_t *ostart, *oend;
|
||||
st_sample_t out[2], tmpOut;
|
||||
|
||||
const int numChannels = stereo ? 2 : 1;
|
||||
int i;
|
||||
|
||||
ostart = obuf;
|
||||
oend = obuf + osamp * 2;
|
||||
|
|
@ -120,16 +138,17 @@ int LinearRateConverter<stereo, reverseStereo>::flow(AudioInputStream &input, st
|
|||
|
||||
// read enough input samples so that ipos > opos
|
||||
while (ipos <= opos) {
|
||||
|
||||
// Abort if we reached the end of the input buffer
|
||||
if (input.eof())
|
||||
goto the_end;
|
||||
|
||||
ilast[0] = icur[0];
|
||||
icur[0] = input.read();
|
||||
if (stereo) {
|
||||
ilast[1] = icur[1];
|
||||
icur[1] = input.read();
|
||||
// Check if we have to refill the buffer
|
||||
if (inLen == 0) {
|
||||
inPtr = inBuf;
|
||||
inLen = input.readBuffer(inBuf, ARRAYSIZE(inBuf));
|
||||
if (inLen <= 0)
|
||||
goto the_end;
|
||||
}
|
||||
for (i = 0; i < numChannels; i++) {
|
||||
ilast[i] = icur[i];
|
||||
icur[i] = *inPtr++;
|
||||
inLen--;
|
||||
}
|
||||
ipos++;
|
||||
}
|
||||
|
|
@ -185,7 +204,10 @@ public:
|
|||
int16 tmp[2];
|
||||
st_size_t len = osamp;
|
||||
assert(input.isStereo() == stereo);
|
||||
while (!input.eof() && len--) {
|
||||
|
||||
// TODO: use readBuffer
|
||||
|
||||
while (!input.eos() && len--) {
|
||||
tmp[0] = tmp[1] = (input.read() * vol) >> 8;
|
||||
if (stereo)
|
||||
tmp[reverseStereo ? 0 : 1] = (input.read() * vol) >> 8;
|
||||
|
|
|
|||
Loading…
Add table
Add a link
Reference in a new issue