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|
/* 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/list.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 --- 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(end.convertToFramerate(getRate())) {
// TODO: This really looks like Timestamp::operator-
_length = Timestamp(_length.secs() - _start.secs(), _length.numberOfFrames() - _start.numberOfFrames(), 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;
}
// TODO: This really looks like Timestamp::operator+
if (_parent->seek(Timestamp(_pos.secs() + _start.secs(), _pos.numberOfFrames() + _start.numberOfFrames(), getRate()))) {
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<bool stereo, bool is16Bit, bool isUnsigned, bool isLE>
class LinearMemoryStream : public SeekableAudioStream {
protected:
const byte *_ptr;
const byte *_end;
const byte *_loopPtr;
const byte *_loopEnd;
const int _rate;
const byte *_origPtr;
const Timestamp _playtime;
uint _numLoops; ///< Number of loops to play
uint _numPlayedLoops; ///< Number of loops which have been played
public:
LinearMemoryStream(int rate, const byte *ptr, uint len, uint loopOffset, uint loopLen, bool autoFreeMemory)
: _ptr(ptr), _end(ptr+len), _loopPtr(0), _loopEnd(0), _rate(rate), _playtime(0, len / (is16Bit ? 2 : 1) / (stereo ? 2 : 1), rate) {
if (loopLen) {
_numLoops = 0;
_loopPtr = _ptr + loopOffset;
_loopEnd = _loopPtr + loopLen;
} else {
_numLoops = 1;
}
_numPlayedLoops = 0;
_origPtr = autoFreeMemory ? ptr : 0;
}
virtual ~LinearMemoryStream() {
free(const_cast<byte *>(_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<bool stereo, bool is16Bit, bool isUnsigned, bool isLE>
int LinearMemoryStream<stereo, is16Bit, isUnsigned, isLE>::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);
// Loop, if looping was specified
// TODO: Handle non-infinite loops
if (_loopPtr && _ptr >= _end) {
_ptr = _loopPtr;
_end = _loopEnd;
_numPlayedLoops++;
}
}
return numSamples-samples;
}
template<bool stereo, bool is16Bit, bool isUnsigned, bool isLE>
bool LinearMemoryStream<stereo, is16Bit, isUnsigned, isLE>::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<bool stereo, bool is16Bit, bool isUnsigned, bool isLE>
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
int _beginLoop; ///< Loop start parameter
int _endLoop; ///< Loop end parameter, currently not implemented
bool _loop; ///< Determines if the stream should be looped when it finishes
uint _numLoops; ///< Number of loops to play
uint _numPlayedLoops; ///< Number of loops which have been played
public:
LinearDiskStream(int rate, uint beginLoop, uint endLoop, bool disposeStream, Common::SeekableReadStream *stream, LinearDiskStreamAudioBlock *block, uint numBlocks, bool loop)
: _rate(rate), _playtime(0, rate), _stream(stream), _beginLoop(beginLoop), _endLoop(endLoop), _disposeAfterUse(disposeStream),
_audioBlockCount(numBlocks), _loop(loop), _numLoops(loop ? 0 : 1), _numPlayedLoops(0) {
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<bool stereo, bool is16Bit, bool isUnsigned, bool isLE>
int LinearDiskStream<stereo, is16Bit, isUnsigned, isLE>::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<int>(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;
}
// Looping
if (_diskLeft == 0 && _loop) {
// Reset the stream
_currentBlock = 0;
_filePos = _audioBlock[_currentBlock].pos + _beginLoop;
_diskLeft = _audioBlock[_currentBlock].len;
_numPlayedLoops++;
}
}
// 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 stereo, bool is16Bit, bool isUnsigned, bool isLE>
bool LinearDiskStream<stereo, is16Bit, isUnsigned, isLE>::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<STEREO, true, UNSIGNED, true>(rate, ptr, len, loopOffset, loopLen, autoFree); \
else \
return new LinearMemoryStream<STEREO, true, UNSIGNED, false>(rate, ptr, len, loopOffset, loopLen, autoFree); \
} else \
return new LinearMemoryStream<STEREO, false, UNSIGNED, false>(rate, ptr, len, loopOffset, loopLen, autoFree)
SeekableAudioStream *makeLinearInputStream(const byte *ptr, uint32 len, int rate, byte flags, uint loopStart, uint loopEnd) {
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;
uint loopOffset = 0, loopLen = 0;
if (flags & Mixer::FLAG_LOOP) {
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((len & 3) == 0 && (loopLen & 3) == 0);
} else if (is16Bit || isStereo) {
assert((len & 1) == 0 && (loopLen & 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);
}
}
}
#define MAKE_LINEAR_DISK(STEREO, UNSIGNED) \
if (is16Bit) { \
if (isLE) \
return new LinearDiskStream<STEREO, true, UNSIGNED, true>(rate, loopStart, loopEnd, takeOwnership, stream, block, numBlocks, loop); \
else \
return new LinearDiskStream<STEREO, true, UNSIGNED, false>(rate, loopStart, loopEnd, takeOwnership, stream, block, numBlocks, loop); \
} else \
return new LinearDiskStream<STEREO, false, UNSIGNED, false>(rate, loopStart, loopEnd, takeOwnership, stream, block, numBlocks, loop)
SeekableAudioStream *makeLinearDiskStream(Common::SeekableReadStream *stream, LinearDiskStreamAudioBlock *block, int numBlocks, int rate, byte flags, bool takeOwnership, uint loopStart, uint loopEnd) {
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 loop = (flags & Mixer::FLAG_LOOP) != 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);
}
}
}
#pragma mark -
#pragma mark --- Appendable audio stream ---
#pragma mark -
struct Buffer {
byte *start;
byte *end;
};
/**
* Wrapped memory stream.
*/
class BaseAppendableMemoryStream : public AppendableAudioStream {
protected:
/**
* A mutex to avoid access problems (causing e.g. corruption of
* the linked list) in thread aware environments.
*/
Common::Mutex _mutex;
// List of all queued buffers
Common::List<Buffer> _bufferQueue;
bool _finalized;
const int _rate;
// Position in the front buffer, if any
byte *_pos;
inline bool eosIntern() const { return _bufferQueue.empty(); };
public:
BaseAppendableMemoryStream(int rate);
~BaseAppendableMemoryStream();
bool endOfStream() const { return _finalized && eosIntern(); }
bool endOfData() const { return eosIntern(); }
int getRate() const { return _rate; }
void finish() { _finalized = true; }
void queueBuffer(byte *data, uint32 size);
};
/**
* Wrapped memory stream.
*/
template<bool stereo, bool is16Bit, bool isUnsigned, bool isLE>
class AppendableMemoryStream : public BaseAppendableMemoryStream {
public:
AppendableMemoryStream(int rate) : BaseAppendableMemoryStream(rate) {}
bool isStereo() const { return stereo; }
int readBuffer(int16 *buffer, const int numSamples);
};
BaseAppendableMemoryStream::BaseAppendableMemoryStream(int rate)
: _finalized(false), _rate(rate), _pos(0) {
}
BaseAppendableMemoryStream::~BaseAppendableMemoryStream() {
// Clear the queue
Common::List<Buffer>::iterator iter;
for (iter = _bufferQueue.begin(); iter != _bufferQueue.end(); ++iter)
delete[] iter->start;
}
template<bool stereo, bool is16Bit, bool isUnsigned, bool isLE>
int AppendableMemoryStream<stereo, is16Bit, isUnsigned, isLE>::readBuffer(int16 *buffer, const int numSamples) {
Common::StackLock lock(_mutex);
int samples = numSamples;
while (samples > 0 && !eosIntern()) {
Buffer buf = *_bufferQueue.begin();
if (_pos == 0)
_pos = buf.start;
assert(buf.start <= _pos && _pos <= buf.end);
const int samplesLeftInCurBuffer = buf.end - _pos;
if (samplesLeftInCurBuffer == 0) {
delete[] buf.start;
_bufferQueue.erase(_bufferQueue.begin());
_pos = 0;
continue;
}
int len = MIN(samples, samplesLeftInCurBuffer / (is16Bit ? 2 : 1));
samples -= len;
do {
*buffer++ = READ_ENDIAN_SAMPLE(is16Bit, isUnsigned, _pos, isLE);
_pos += (is16Bit ? 2 : 1);
} while (--len);
}
return numSamples - samples;
}
void BaseAppendableMemoryStream::queueBuffer(byte *data, uint32 size) {
Common::StackLock lock(_mutex);
/*
// Verify the buffer size is sane
if (is16Bit && stereo) {
assert((size & 3) == 0);
} else if (is16Bit || stereo) {
assert((size & 1) == 0);
}
*/
// Verify that the stream has not yet been finalized (by a call to finish())
assert(!_finalized);
// Queue the buffer
Buffer buf = {data, data+size};
_bufferQueue.push_back(buf);
#if 0
// Output some stats
uint totalSize = 0;
Common::List<Buffer>::iterator iter;
for (iter = _bufferQueue.begin(); iter != _bufferQueue.end(); ++iter)
totalSize += iter->end - iter->start;
printf("AppendableMemoryStream::queueBuffer: added a %d byte buf, a total of %d bytes are queued\n",
size, totalSize);
#endif
}
#define MAKE_WRAPPED(STEREO, UNSIGNED) \
if (is16Bit) { \
if (isLE) \
return new AppendableMemoryStream<STEREO, true, UNSIGNED, true>(rate); \
else \
return new AppendableMemoryStream<STEREO, true, UNSIGNED, false>(rate); \
} else \
return new AppendableMemoryStream<STEREO, false, UNSIGNED, false>(rate)
AppendableAudioStream *makeAppendableAudioStream(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;
if (isStereo) {
if (isUnsigned) {
MAKE_WRAPPED(true, true);
} else {
MAKE_WRAPPED(true, false);
}
} else {
if (isUnsigned) {
MAKE_WRAPPED(false, true);
} else {
MAKE_WRAPPED(false, false);
}
}
}
#pragma mark -
#pragma mark --- Appendable 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<StreamHolder> _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
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