/* ScummVM - Graphic Adventure Engine * * ScummVM is the legal property of its developers, whose names * are too numerous to list here. Please refer to the COPYRIGHT * file distributed with this source distribution. * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License * as published by the Free Software Foundation; either version 2 * of the License, or (at your option) any later version. * This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA. * * $URL$ * $Id$ * */ #include "common/debug.h" #include "common/endian.h" #include "common/file.h" #include "common/queue.h" #include "common/util.h" #include "sound/audiostream.h" #include "sound/mixer.h" #include "sound/mp3.h" #include "sound/vorbis.h" #include "sound/flac.h" // This used to be an inline template function, but // buggy template function handling in MSVC6 forced // us to go with the macro approach. So far this is // the only template function that MSVC6 seemed to // compile incorrectly. Knock on wood. #define READ_ENDIAN_SAMPLE(is16Bit, isUnsigned, ptr, isLE) \ ((is16Bit ? (isLE ? READ_LE_UINT16(ptr) : READ_BE_UINT16(ptr)) : (*ptr << 8)) ^ (isUnsigned ? 0x8000 : 0)) namespace Audio { struct StreamFileFormat { /** Decodername */ const char* decoderName; const char* fileExtension; /** * Pointer to a function which tries to open a file of type StreamFormat. * Return NULL in case of an error (invalid/nonexisting file). */ SeekableAudioStream *(*openStreamFile)(Common::SeekableReadStream *stream, bool disposeAfterUse); }; static const StreamFileFormat STREAM_FILEFORMATS[] = { /* decoderName, fileExt, openStreamFuntion */ #ifdef USE_FLAC { "Flac", ".flac", makeFlacStream }, { "Flac", ".fla", makeFlacStream }, #endif #ifdef USE_VORBIS { "Ogg Vorbis", ".ogg", makeVorbisStream }, #endif #ifdef USE_MAD { "MPEG Layer 3", ".mp3", makeMP3Stream }, #endif { NULL, NULL, NULL } // Terminator }; SeekableAudioStream *SeekableAudioStream::openStreamFile(const Common::String &basename) { SeekableAudioStream *stream = NULL; Common::File *fileHandle = new Common::File(); for (int i = 0; i < ARRAYSIZE(STREAM_FILEFORMATS)-1 && stream == NULL; ++i) { Common::String filename = basename + STREAM_FILEFORMATS[i].fileExtension; fileHandle->open(filename); if (fileHandle->isOpen()) { // Create the stream object stream = STREAM_FILEFORMATS[i].openStreamFile(fileHandle, true); fileHandle = 0; break; } } delete fileHandle; if (stream == NULL) { debug(1, "AudioStream: Could not open compressed AudioFile %s", basename.c_str()); } return stream; } #pragma mark - #pragma mark --- LoopingAudioStream --- #pragma mark - LoopingAudioStream::LoopingAudioStream(RewindableAudioStream *stream, uint loops, bool disposeAfterUse) : _parent(stream), _disposeAfterUse(disposeAfterUse), _loops(loops), _completeIterations(0) { } LoopingAudioStream::~LoopingAudioStream() { if (_disposeAfterUse) delete _parent; } int LoopingAudioStream::readBuffer(int16 *buffer, const int numSamples) { int samplesRead = _parent->readBuffer(buffer, numSamples); if (_parent->endOfStream()) { ++_completeIterations; if (_completeIterations == _loops) return samplesRead; int remainingSamples = numSamples - samplesRead; if (!_parent->rewind()) { // TODO: Properly indicate error _loops = _completeIterations = 1; return samplesRead; } samplesRead += _parent->readBuffer(buffer + samplesRead, remainingSamples); } return samplesRead; } bool LoopingAudioStream::endOfData() const { return (_loops != 0 && (_completeIterations == _loops)); } AudioStream *makeLoopingAudioStream(RewindableAudioStream *stream, uint loops) { if (loops != 1) return new LoopingAudioStream(stream, loops); else return stream; } AudioStream *makeLoopingAudioStream(SeekableAudioStream *stream, Timestamp start, Timestamp end, uint loops) { if (!start.totalNumberOfFrames() && (!end.totalNumberOfFrames() || end == stream->getLength())) { return makeLoopingAudioStream(stream, loops); } else { if (!end.totalNumberOfFrames()) end = stream->getLength(); if (start >= end) { warning("makeLoopingAudioStream: start (%d) >= end (%d)", start.msecs(), end.msecs()); delete stream; return 0; } return makeLoopingAudioStream(new SubSeekableAudioStream(stream, start, end), loops); } } #pragma mark - #pragma mark --- SubLoopingAudioStream --- #pragma mark - SubLoopingAudioStream::SubLoopingAudioStream(SeekableAudioStream *stream, uint loops, const Timestamp loopStart, const Timestamp loopEnd, bool disposeAfterUse) : _parent(stream), _disposeAfterUse(disposeAfterUse), _loops(loops), _pos(0, getRate() * (isStereo() ? 2 : 1)), _loopStart(loopStart.convertToFramerate(getRate() * (isStereo() ? 2 : 1))), _loopEnd(loopEnd.convertToFramerate(getRate() * (isStereo() ? 2 : 1))), _done(false) { if (!_parent->rewind()) _done = true; } SubLoopingAudioStream::~SubLoopingAudioStream() { if (_disposeAfterUse) delete _parent; } int SubLoopingAudioStream::readBuffer(int16 *buffer, const int numSamples) { int framesLeft = MIN(_loopEnd.frameDiff(_pos), numSamples); int framesRead = _parent->readBuffer(buffer, framesLeft); _pos = _pos.addFrames(framesRead); if (framesLeft < numSamples || framesRead < framesLeft) { if (_loops != 0) { --_loops; if (!_loops) { _done = true; return framesRead; } } if (!_parent->seek(_loopStart)) { _done = true; return framesRead; } _pos = _loopStart; framesLeft = numSamples - framesLeft; framesRead += _parent->readBuffer(buffer + framesRead, framesLeft); if (_parent->endOfStream()) _done = true; } return framesRead; } #pragma mark - #pragma mark --- SubSeekableAudioStream --- #pragma mark - SubSeekableAudioStream::SubSeekableAudioStream(SeekableAudioStream *parent, const Timestamp start, const Timestamp end, bool disposeAfterUse) : _parent(parent), _disposeAfterUse(disposeAfterUse), _start(start.convertToFramerate(getRate())), _pos(0, getRate() * (isStereo() ? 2 : 1)), _length((start - end).convertToFramerate(getRate())) { _parent->seek(_start); } SubSeekableAudioStream::~SubSeekableAudioStream() { if (_disposeAfterUse) delete _parent; } int SubSeekableAudioStream::readBuffer(int16 *buffer, const int numSamples) { int framesLeft = MIN(_length.frameDiff(_pos), numSamples); int framesRead = _parent->readBuffer(buffer, framesLeft); _pos = _pos.addFrames(framesRead); return framesRead; } bool SubSeekableAudioStream::seek(const Timestamp &where) { _pos = where.convertToFramerate(getRate()); if (_pos > _length) { _pos = _length; return false; } if (_parent->seek(_pos + _start)) { return true; } else { _pos = _length; return false; } } #pragma mark - #pragma mark --- LinearMemoryStream --- #pragma mark - uint32 calculateSampleOffset(const Timestamp &where, int rate) { return where.convertToFramerate(rate).totalNumberOfFrames(); } /** * A simple raw audio stream, purely memory based. It operates on a single * block of data, which is passed to it upon creation. * Optionally supports looping the sound. * * Design note: This code tries to be as efficient as possible (without * resorting to assembly, that is). To this end, it is written as a template * class. This way the compiler can create optimized code for each special * case. This results in a total of 12 versions of the code being generated. */ template class LinearMemoryStream : public SeekableAudioStream { protected: const byte *_ptr; const byte *_end; const int _rate; const byte *_origPtr; const bool _disposeAfterUse; const Timestamp _playtime; public: LinearMemoryStream(int rate, const byte *ptr, uint len, bool autoFreeMemory) : _ptr(ptr), _end(ptr+len), _rate(rate), _origPtr(ptr), _disposeAfterUse(autoFreeMemory), _playtime(0, len / (is16Bit ? 2 : 1) / (stereo ? 2 : 1), rate) { } virtual ~LinearMemoryStream() { if (_disposeAfterUse) free(const_cast(_origPtr)); } int readBuffer(int16 *buffer, const int numSamples); bool isStereo() const { return stereo; } bool endOfData() const { return _ptr >= _end; } int getRate() const { return _rate; } bool seek(const Timestamp &where); Timestamp getLength() const { return _playtime; } }; template int LinearMemoryStream::readBuffer(int16 *buffer, const int numSamples) { int samples = numSamples; while (samples > 0 && _ptr < _end) { int len = MIN(samples, (int)(_end - _ptr) / (is16Bit ? 2 : 1)); samples -= len; do { *buffer++ = READ_ENDIAN_SAMPLE(is16Bit, isUnsigned, _ptr, isLE); _ptr += (is16Bit ? 2 : 1); } while (--len); } return numSamples-samples; } template bool LinearMemoryStream::seek(const Timestamp &where) { const uint8 *ptr = _origPtr + calculateSampleOffset(where, getRate()) * (is16Bit ? 2 : 1) * (stereo ? 2 : 1); if (ptr > _end) { _ptr = _end; return false; } else if (ptr == _end) { _ptr = _end; return true; } else { _ptr = ptr; return true; } } #pragma mark - #pragma mark --- LinearDiskStream --- #pragma mark - /** * LinearDiskStream. This can stream linear (PCM) audio from disk. The * function takes an pointer to an array of LinearDiskStreamAudioBlock which defines the * start position and length of each block of uncompressed audio in the stream. */ template class LinearDiskStream : public SeekableAudioStream { // Allow backends to override buffer size #ifdef CUSTOM_AUDIO_BUFFER_SIZE static const int32 BUFFER_SIZE = CUSTOM_AUDIO_BUFFER_SIZE; #else static const int32 BUFFER_SIZE = 16384; #endif protected: byte* _buffer; ///< Streaming buffer const byte *_ptr; ///< Pointer to current position in stream buffer const int _rate; ///< Sample rate of stream Timestamp _playtime; ///< Calculated total play time Common::SeekableReadStream *_stream; ///< Stream to read data from int32 _filePos; ///< Current position in stream int32 _diskLeft; ///< Samples left in stream in current block not yet read to buffer int32 _bufferLeft; ///< Samples left in buffer in current block bool _disposeAfterUse; ///< If true, delete stream object when LinearDiskStream is destructed LinearDiskStreamAudioBlock *_audioBlock; ///< Audio block list int _audioBlockCount; ///< Number of blocks in _audioBlock int _currentBlock; ///< Current audio block number public: LinearDiskStream(int rate, bool disposeStream, Common::SeekableReadStream *stream, LinearDiskStreamAudioBlock *block, uint numBlocks) : _rate(rate), _playtime(0, rate), _stream(stream), _disposeAfterUse(disposeStream), _audioBlockCount(numBlocks) { assert(numBlocks > 0); // Allocate streaming buffer if (is16Bit) { _buffer = (byte *)malloc(BUFFER_SIZE * sizeof(int16)); } else { _buffer = (byte *)malloc(BUFFER_SIZE * sizeof(byte)); } _ptr = _buffer; _bufferLeft = 0; // Copy audio block data to our buffer // TODO: Replace this with a Common::Array or Common::List to // make it a little friendlier. _audioBlock = new LinearDiskStreamAudioBlock[numBlocks]; memcpy(_audioBlock, block, numBlocks * sizeof(LinearDiskStreamAudioBlock)); // Set current buffer state, playing first block _currentBlock = 0; _filePos = _audioBlock[_currentBlock].pos; _diskLeft = _audioBlock[_currentBlock].len; // Add up length of all blocks in order to caluclate total play time int len = 0; for (int r = 0; r < _audioBlockCount; r++) { len += _audioBlock[r].len; } _playtime = Timestamp(0, len / (is16Bit ? 2 : 1) / (stereo ? 2 : 1), rate); } virtual ~LinearDiskStream() { if (_disposeAfterUse) { delete _stream; } delete[] _audioBlock; free(_buffer); } int readBuffer(int16 *buffer, const int numSamples); bool isStereo() const { return stereo; } bool endOfData() const { return (_currentBlock == _audioBlockCount - 1) && (_diskLeft == 0) && (_bufferLeft == 0); } int getRate() const { return _rate; } Timestamp getLength() const { return _playtime; } bool seek(const Timestamp &where); }; template int LinearDiskStream::readBuffer(int16 *buffer, const int numSamples) { int oldPos = _stream->pos(); bool restoreFilePosition = false; int samples = numSamples; while (samples > 0 && ((_diskLeft > 0 || _bufferLeft > 0) || (_currentBlock != _audioBlockCount - 1)) ) { // Output samples in the buffer to the output int len = MIN(samples, _bufferLeft); samples -= len; _bufferLeft -= len; while (len > 0) { *buffer++ = READ_ENDIAN_SAMPLE(is16Bit, isUnsigned, _ptr, isLE); _ptr += (is16Bit ? 2 : 1); len--; } // Have we now finished this block? If so, read the next block if ((_bufferLeft == 0) && (_diskLeft == 0) && (_currentBlock != _audioBlockCount - 1)) { // Next block _currentBlock++; _filePos = _audioBlock[_currentBlock].pos; _diskLeft = _audioBlock[_currentBlock].len; } // Now read more data from disk if there is more to be read if ((_bufferLeft == 0) && (_diskLeft > 0)) { int32 readAmount = MIN(_diskLeft, BUFFER_SIZE); _stream->seek(_filePos, SEEK_SET); _stream->read(_buffer, readAmount * (is16Bit? 2: 1)); // Amount of data in buffer is now the amount read in, and // the amount left to read on disk is decreased by the same amount _bufferLeft = readAmount; _diskLeft -= readAmount; _ptr = (byte *)_buffer; _filePos += readAmount * (is16Bit ? 2 : 1); // Set this flag now we've used the file, it restores it's // original position. restoreFilePosition = true; } } // In case calling code relies on the position of this stream staying // constant, I restore the location if I've changed it. This is probably // not necessary. if (restoreFilePosition) { _stream->seek(oldPos, SEEK_SET); } return numSamples - samples; } template bool LinearDiskStream::seek(const Timestamp &where) { const uint32 seekSample = calculateSampleOffset(where, getRate()) * (stereo ? 2 : 1); uint32 curSample = 0; // Search for the disk block in which the specific sample is placed _currentBlock = 0; while (_currentBlock < _audioBlockCount) { uint32 nextBlockSample = curSample + _audioBlock[_currentBlock].len; if (nextBlockSample > seekSample) break; curSample = nextBlockSample; ++_currentBlock; } _filePos = 0; _diskLeft = 0; _bufferLeft = 0; if (_currentBlock == _audioBlockCount) { return ((seekSample - curSample) == (uint32)_audioBlock[_currentBlock - 1].len); } else { const uint32 offset = seekSample - curSample; _filePos = _audioBlock[_currentBlock].pos + offset * (is16Bit ? 2 : 1); _diskLeft = _audioBlock[_currentBlock].len - offset; return true; } } #pragma mark - #pragma mark --- Input stream factory --- #pragma mark - /* In the following, we use preprocessor / macro tricks to simplify the code * which instantiates the input streams. We used to use template functions for * this, but MSVC6 / EVC 3-4 (used for WinCE builds) are extremely buggy when it * comes to this feature of C++... so as a compromise we use macros to cut down * on the (source) code duplication a bit. * So while normally macro tricks are said to make maintenance harder, in this * particular case it should actually help it :-) */ #define MAKE_LINEAR(STEREO, UNSIGNED) \ if (is16Bit) { \ if (isLE) \ return new LinearMemoryStream(rate, ptr, len, autoFree); \ else \ return new LinearMemoryStream(rate, ptr, len, autoFree); \ } else \ return new LinearMemoryStream(rate, ptr, len, autoFree) SeekableAudioStream *makeLinearInputStream(const byte *ptr, uint32 len, int rate, byte flags) { const bool isStereo = (flags & Mixer::FLAG_STEREO) != 0; const bool is16Bit = (flags & Mixer::FLAG_16BITS) != 0; const bool isUnsigned = (flags & Mixer::FLAG_UNSIGNED) != 0; const bool isLE = (flags & Mixer::FLAG_LITTLE_ENDIAN) != 0; const bool autoFree = (flags & Mixer::FLAG_AUTOFREE) != 0; // Verify the buffer sizes are sane if (is16Bit && isStereo) { assert((len & 3) == 0); } else if (is16Bit || isStereo) { assert((len & 1) == 0); } if (isStereo) { if (isUnsigned) { MAKE_LINEAR(true, true); } else { MAKE_LINEAR(true, false); } } else { if (isUnsigned) { MAKE_LINEAR(false, true); } else { MAKE_LINEAR(false, false); } } } AudioStream *makeLinearInputStream(const byte *ptr, uint32 len, int rate, byte flags, uint loopStart, uint loopEnd) { SeekableAudioStream *stream = makeLinearInputStream(ptr, len, rate, flags); const bool isStereo = (flags & Mixer::FLAG_STEREO) != 0; const bool is16Bit = (flags & Mixer::FLAG_16BITS) != 0; const bool isLooping = (flags & Mixer::FLAG_LOOP) != 0; if (isLooping) { uint loopOffset = 0, loopLen = 0; if (loopEnd == 0) loopEnd = len; assert(loopStart <= loopEnd); assert(loopEnd <= len); loopOffset = loopStart; loopLen = loopEnd - loopStart; // Verify the buffer sizes are sane if (is16Bit && isStereo) assert((loopLen & 3) == 0 && (loopStart & 3) == 0 && (loopEnd & 3) == 0); else if (is16Bit || isStereo) assert((loopLen & 1) == 0 && (loopStart & 1) == 0 && (loopEnd & 1) == 0); const uint32 extRate = stream->getRate() * (is16Bit ? 2 : 1) * (isStereo ? 2 : 1); return new SubLoopingAudioStream(stream, 0, Timestamp(0, loopStart, extRate), Timestamp(0, loopEnd, extRate)); } else { return stream; } } #define MAKE_LINEAR_DISK(STEREO, UNSIGNED) \ if (is16Bit) { \ if (isLE) \ return new LinearDiskStream(rate, takeOwnership, stream, block, numBlocks); \ else \ return new LinearDiskStream(rate, takeOwnership, stream, block, numBlocks); \ } else \ return new LinearDiskStream(rate, takeOwnership, stream, block, numBlocks) SeekableAudioStream *makeLinearDiskStream(Common::SeekableReadStream *stream, LinearDiskStreamAudioBlock *block, int numBlocks, int rate, byte flags, bool takeOwnership) { const bool isStereo = (flags & Mixer::FLAG_STEREO) != 0; const bool is16Bit = (flags & Mixer::FLAG_16BITS) != 0; const bool isUnsigned = (flags & Mixer::FLAG_UNSIGNED) != 0; const bool isLE = (flags & Mixer::FLAG_LITTLE_ENDIAN) != 0; if (isStereo) { if (isUnsigned) { MAKE_LINEAR_DISK(true, true); } else { MAKE_LINEAR_DISK(true, false); } } else { if (isUnsigned) { MAKE_LINEAR_DISK(false, true); } else { MAKE_LINEAR_DISK(false, false); } } } AudioStream *makeLinearDiskStream(Common::SeekableReadStream *stream, LinearDiskStreamAudioBlock *block, int numBlocks, int rate, byte flags, bool disposeStream, uint loopStart, uint loopEnd) { SeekableAudioStream *s = makeLinearDiskStream(stream, block, numBlocks, rate, flags, disposeStream); const bool isStereo = (flags & Mixer::FLAG_STEREO) != 0; const bool is16Bit = (flags & Mixer::FLAG_16BITS) != 0; const bool isLooping = (flags & Mixer::FLAG_LOOP) != 0; if (isLooping) { uint loopOffset = 0, loopLen = 0; const uint len = s->getLength().totalNumberOfFrames() / (is16Bit ? 2 : 1) / (isStereo ? 2 : 1); if (loopEnd == 0) loopEnd = len; assert(loopStart <= loopEnd); assert(loopEnd <= len); loopOffset = loopStart; loopLen = loopEnd - loopStart; // Verify the buffer sizes are sane if (is16Bit && isStereo) assert((loopLen & 3) == 0 && (loopStart & 3) == 0 && (loopEnd & 3) == 0); else if (is16Bit || isStereo) assert((loopLen & 1) == 0 && (loopStart & 1) == 0 && (loopEnd & 3) == 0); const uint32 extRate = s->getRate() * (is16Bit ? 2 : 1) * (isStereo ? 2 : 1); return new SubLoopingAudioStream(s, 0, Timestamp(0, loopStart, extRate), Timestamp(0, loopEnd, extRate)); } else { return s; } } #pragma mark - #pragma mark --- Queueing audio stream --- #pragma mark - class QueuingAudioStreamImpl : public QueuingAudioStream { private: /** * We queue a number of (pointers to) audio stream objects. * In addition, we need to remember for each stream whether * to dispose it after all data has been read from it. * Hence, we don't store pointers to stream objects directly, * but rather StreamHolder structs. */ struct StreamHolder { AudioStream *_stream; bool _disposeAfterUse; StreamHolder(AudioStream *stream, bool disposeAfterUse) : _stream(stream), _disposeAfterUse(disposeAfterUse) {} }; /** * The sampling rate of this audio stream. */ const int _rate; /** * Whether this audio stream is mono (=false) or stereo (=true). */ const int _stereo; /** * This flag is set by the finish() method only. See there for more details. */ bool _finished; /** * A mutex to avoid access problems (causing e.g. corruption of * the linked list) in thread aware environments. */ Common::Mutex _mutex; /** * The queue of audio streams. */ Common::Queue _queue; public: QueuingAudioStreamImpl(int rate, bool stereo) : _rate(rate), _stereo(stereo), _finished(false) {} ~QueuingAudioStreamImpl(); // Implement the AudioStream API virtual int readBuffer(int16 *buffer, const int numSamples); virtual bool isStereo() const { return _stereo; } virtual int getRate() const { return _rate; } virtual bool endOfData() const { //Common::StackLock lock(_mutex); return _queue.empty(); } virtual bool endOfStream() const { return _finished; } // Implement the QueuingAudioStream API virtual void queueAudioStream(AudioStream *stream, bool disposeAfterUse); virtual void finish() { _finished = true; } uint32 numQueuedStreams() const { //Common::StackLock lock(_mutex); return _queue.size(); } }; QueuingAudioStreamImpl::~QueuingAudioStreamImpl() { while (!_queue.empty()) { StreamHolder tmp = _queue.pop(); if (tmp._disposeAfterUse) delete tmp._stream; } } void QueuingAudioStreamImpl::queueAudioStream(AudioStream *stream, bool disposeAfterUse) { if ((stream->getRate() != getRate()) || (stream->isStereo() != isStereo())) error("QueuingAudioStreamImpl::queueAudioStream: stream has mismatched parameters"); Common::StackLock lock(_mutex); _queue.push(StreamHolder(stream, disposeAfterUse)); } int QueuingAudioStreamImpl::readBuffer(int16 *buffer, const int numSamples) { Common::StackLock lock(_mutex); int samplesDecoded = 0; while (samplesDecoded < numSamples && !_queue.empty()) { AudioStream *stream = _queue.front()._stream; samplesDecoded += stream->readBuffer(buffer + samplesDecoded, numSamples - samplesDecoded); if (stream->endOfData() ) { StreamHolder tmp = _queue.pop(); if (tmp._disposeAfterUse) delete stream; } } return samplesDecoded; } QueuingAudioStream *makeQueuingAudioStream(int rate, bool stereo) { return new QueuingAudioStreamImpl(rate, stereo); } } // End of namespace Audio