/* ScummVM - Scumm Interpreter * Copyright (C) 2001-2006 The ScummVM project * * 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/stdafx.h" #include "common/endian.h" #include "common/file.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" 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). */ AudioStream* (*openStreamFile)(Common::SeekableReadStream *stream, bool disposeAfterUse, uint32 startTime, uint32 duration, uint numLoops); }; 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 }; AudioStream* AudioStream::openStreamFile(const char *filename) { char buffer[1024]; const uint len = strlen(filename); assert(len+6 < sizeof(buffer)); // we need a bigger buffer if wrong memcpy(buffer, filename, len); buffer[len] = '.'; char *ext = &buffer[len+1]; AudioStream* stream = NULL; Common::File *fileHandle = new Common::File(); for (int i = 0; i < ARRAYSIZE(STREAM_FILEFORMATS)-1 && stream == NULL; ++i) { strcpy(ext, STREAM_FILEFORMATS[i].fileExtension); fileHandle->open(buffer); if (fileHandle->isOpen()) { stream = STREAM_FILEFORMATS[i].openStreamFile(fileHandle, true, 0, 0, 1); fileHandle = 0; break; } } delete fileHandle; if (stream == NULL) { debug(1, "AudioStream: Could not open compressed AudioFile %s", filename); } return stream; } #pragma mark - #pragma mark --- LinearMemoryStream --- #pragma mark - /** * 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 optimized as possible (without * resorting to assembly, that is). To this end, it is written as a template * class. This way the compiler can actually create optimized code for each * special code. This results in a total of 12 versions of the code being * generated. */ template class LinearMemoryStream : public AudioStream { protected: const byte *_ptr; const byte *_end; const byte *_loopPtr; const byte *_loopEnd; const int _rate; const byte *_origPtr; inline bool eosIntern() const { return _ptr >= _end; }; 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) { // Verify the buffer sizes are sane if (is16Bit && stereo) assert((len & 3) == 0 && (loopLen & 3) == 0); else if (is16Bit || stereo) assert((len & 1) == 0 && (loopLen & 1) == 0); if (loopLen) { _loopPtr = _ptr + loopOffset; _loopEnd = _loopPtr + loopLen; } if (stereo) // Stereo requires even sized data assert(len % 2 == 0); _origPtr = autoFreeMemory ? ptr : 0; } ~LinearMemoryStream() { free(const_cast(_origPtr)); } int readBuffer(int16 *buffer, const int numSamples); bool isStereo() const { return stereo; } bool endOfData() const { return eosIntern(); } int getRate() const { return _rate; } }; template int LinearMemoryStream::readBuffer(int16 *buffer, const int numSamples) { int samples = 0; while (samples < numSamples && !eosIntern()) { const int len = MIN(numSamples, samples + (int)(_end - _ptr) / (is16Bit ? 2 : 1)); while (samples < len) { *buffer++ = READ_ENDIAN_SAMPLE(is16Bit, isUnsigned, _ptr, isLE); _ptr += (is16Bit ? 2 : 1); samples++; } // Loop, if looping was specified if (_loopPtr && eosIntern()) { _ptr = _loopPtr; _end = _loopEnd; } } return samples; } #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, loopOffset, loopLen, autoFree); \ else \ return new LinearMemoryStream(rate, ptr, len, loopOffset, loopLen, autoFree); \ } else \ return new LinearMemoryStream(rate, ptr, len, loopOffset, loopLen, autoFree) AudioStream *makeLinearInputStream(int rate, byte flags, const byte *ptr, uint32 len, uint loopOffset, uint loopLen) { const bool isStereo = (flags & Audio::Mixer::FLAG_STEREO) != 0; const bool is16Bit = (flags & Audio::Mixer::FLAG_16BITS) != 0; const bool isUnsigned = (flags & Audio::Mixer::FLAG_UNSIGNED) != 0; const bool isLE = (flags & Audio::Mixer::FLAG_LITTLE_ENDIAN) != 0; const bool autoFree = (flags & Audio::Mixer::FLAG_AUTOFREE) != 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); } } } #pragma mark - #pragma mark --- Appendable audio stream --- #pragma mark - /** * Wrapped memory stream. */ template class AppendableMemoryStream : public AppendableAudioStream { protected: Common::Mutex _mutex; byte *_bufferStart; byte *_bufferEnd; byte *_pos; byte *_end; bool _finalized; const int _rate; inline bool eosIntern() const { return _end == _pos; }; public: AppendableMemoryStream(int rate, uint bufferSize); ~AppendableMemoryStream(); int readBuffer(int16 *buffer, const int numSamples); bool isStereo() const { return stereo; } bool endOfStream() const { return _finalized && eosIntern(); } bool endOfData() const { return eosIntern(); } int getRate() const { return _rate; } void append(const byte *data, uint32 len); void finish() { _finalized = true; } }; template AppendableMemoryStream::AppendableMemoryStream(int rate, uint bufferSize) : _finalized(false), _rate(rate) { // Verify the buffer size is sane if (is16Bit && stereo) assert((bufferSize & 3) == 0); else if (is16Bit || stereo) assert((bufferSize & 1) == 0); _bufferStart = (byte *)malloc(bufferSize); assert(_bufferStart != NULL); _pos = _end = _bufferStart; _bufferEnd = _bufferStart + bufferSize; } template AppendableMemoryStream::~AppendableMemoryStream() { free(_bufferStart); } template int AppendableMemoryStream::readBuffer(int16 *buffer, const int numSamples) { Common::StackLock lock(_mutex); int samples = 0; while (samples < numSamples && !eosIntern()) { // Wrap around? if (_pos >= _bufferEnd) _pos = _pos - (_bufferEnd - _bufferStart); const byte *endMarker = (_pos > _end) ? _bufferEnd : _end; const int len = MIN(numSamples, samples + (int)(endMarker - _pos) / (is16Bit ? 2 : 1)); while (samples < len) { *buffer++ = READ_ENDIAN_SAMPLE(is16Bit, isUnsigned, _pos, isLE); _pos += (is16Bit ? 2 : 1); samples++; } } return samples; } template void AppendableMemoryStream::append(const byte *data, uint32 len) { Common::StackLock lock(_mutex); // Verify the buffer size is sane if (is16Bit && stereo) assert((len & 3) == 0); else if (is16Bit || stereo) assert((len & 1) == 0); // Verify that the stream has not yet been finalized (by a call to finish()) assert(!_finalized); if (_end + len > _bufferEnd) { // Wrap-around case uint32 size_to_end_of_buffer = _bufferEnd - _end; len -= size_to_end_of_buffer; if ((_end < _pos) || (_bufferStart + len >= _pos)) { debug(2, "AppendableMemoryStream: buffer overflow (A)"); return; } memcpy(_end, data, size_to_end_of_buffer); memcpy(_bufferStart, data + size_to_end_of_buffer, len); _end = _bufferStart + len; } else { if ((_end < _pos) && (_end + len >= _pos)) { debug(2, "AppendableMemoryStream: buffer overflow (B)"); return; } memcpy(_end, data, len); _end += len; } } #define MAKE_WRAPPED(STEREO, UNSIGNED) \ if (is16Bit) { \ if (isLE) \ return new AppendableMemoryStream(rate, len); \ else \ return new AppendableMemoryStream(rate, len); \ } else \ return new AppendableMemoryStream(rate, len) AppendableAudioStream *makeAppendableAudioStream(int rate, byte _flags, uint32 len) { const bool isStereo = (_flags & Audio::Mixer::FLAG_STEREO) != 0; const bool is16Bit = (_flags & Audio::Mixer::FLAG_16BITS) != 0; const bool isUnsigned = (_flags & Audio::Mixer::FLAG_UNSIGNED) != 0; const bool isLE = (_flags & Audio::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); } } } } // End of namespace Audio