/* ScummVM - Scumm Interpreter * Copyright (C) 2001-2003 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., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. * * $Header$ * */ #include "stdafx.h" #include "audiostream.h" #include "mixer.h" #include "base/engine.h" #include "common/file.h" #include "common/util.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 READSAMPLE(is16Bit, isUnsigned, ptr) \ ((is16Bit ? READ_BE_UINT16(ptr) : (*ptr << 8)) ^ (isUnsigned ? 0x8000 : 0)) #define READ_ENDIAN_SAMPLE(is16Bit, isUnsigned, ptr, isLE) \ ((is16Bit ? (isLE ? READ_LE_UINT16(ptr) : READ_BE_UINT16(ptr)) : (*ptr << 8)) ^ (isUnsigned ? 0x8000 : 0)) #pragma mark - #pragma mark --- LinearMemoryStream --- #pragma mark - template class LinearMemoryStream : public AudioInputStream { protected: const byte *_ptr; const byte *_end; const byte *_loopPtr; const byte *_loopEnd; inline bool eosIntern() const { return _ptr >= _end; }; public: LinearMemoryStream(const byte *ptr, uint len, uint loopOffset, uint loopLen) : _ptr(ptr), _end(ptr+len), _loopPtr(0), _loopEnd(0) { // 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); } int readBuffer(int16 *buffer, const int numSamples); int16 read() { //assert(_ptr < _end); int16 val = READ_ENDIAN_SAMPLE(is16Bit, isUnsigned, _ptr, isLE); _ptr += (is16Bit ? 2 : 1); if (_loopPtr && eosIntern()) { _ptr = _loopPtr; _end = _loopEnd; } return val; } bool eos() const { return eosIntern(); } bool isStereo() const { return stereo; } }; 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 --- WrappedMemoryStream --- #pragma mark - // Wrapped memory stream, to be used by the ChannelStream class (and possibly others?) template class WrappedMemoryStream : public WrappedAudioInputStream { protected: byte *_bufferStart; byte *_bufferEnd; byte *_pos; byte *_end; inline bool eosIntern() const { return _end == _pos; }; public: WrappedMemoryStream(uint bufferSize); ~WrappedMemoryStream() { free(_bufferStart); } int readBuffer(int16 *buffer, const int numSamples); int16 read(); bool eos() const { return eosIntern(); } bool isStereo() const { return stereo; } void append(const byte *data, uint32 len); }; template WrappedMemoryStream::WrappedMemoryStream(uint bufferSize) { // 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); _pos = _end = _bufferStart; _bufferEnd = _bufferStart + bufferSize; } template inline int16 WrappedMemoryStream::read() { if (eosIntern()) { // If the stream contains no more data, it is silent... return 0; } int16 val = READSAMPLE(is16Bit, isUnsigned, _pos); _pos += (is16Bit ? 2 : 1); // Wrap around? if (_pos >= _bufferEnd) _pos = _pos - (_bufferEnd - _bufferStart); return val; } template int WrappedMemoryStream::readBuffer(int16 *buffer, const int numSamples) { int samples = 0; while (samples < numSamples && !eosIntern()) { const byte *endMarker = (_pos > _end) ? _bufferEnd : _end; const int len = MIN(numSamples, samples + (int)(endMarker - _pos) / (is16Bit ? 2 : 1)); while (samples < len) { *buffer++ = READSAMPLE(is16Bit, isUnsigned, _pos); _pos += (is16Bit ? 2 : 1); samples++; } // Wrap around? if (_pos >= _bufferEnd) _pos = _pos - (_bufferEnd - _bufferStart); } return samples; } template void WrappedMemoryStream::append(const byte *data, uint32 len) { // Verify the buffer size is sane if (is16Bit && stereo) assert((len & 3) == 0); else if (is16Bit || stereo) assert((len & 1) == 0); 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, "WrappedMemoryStream: 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, "WrappedMemoryStream: buffer overflow (B)"); return; } memcpy(_end, data, len); _end += len; } } #pragma mark - #pragma mark --- MP3 (MAD) stream --- #pragma mark - #ifdef USE_MAD class MP3InputStream : public MusicStream { struct mad_stream _stream; struct mad_frame _frame; struct mad_synth _synth; mad_timer_t _duration; uint32 _posInFrame; uint32 _bufferSize; int _size; bool _isStereo; int _curChannel; File *_file; byte *_ptr; bool init(); void refill(bool first = false); inline bool eosIntern() const; public: MP3InputStream(File *file, mad_timer_t duration, uint size = 0); ~MP3InputStream(); int readBuffer(int16 *buffer, const int numSamples); int16 read(); bool eos() const { return eosIntern(); } bool isStereo() const { return _isStereo; } int getRate() const { return _frame.header.samplerate; } }; /** * Playback the MP3 data in the given file for the specified duration. * * @param file file containing the MP3 data * @param duration playback duration in frames (1/75th of a second), 0 means * playback until EOF * @param size optional, if non-zero this limits playback based on the * number of input bytes rather then a duration */ MP3InputStream::MP3InputStream(File *file, mad_timer_t duration, uint size) { // duration == 0 means: play everything till end of file mad_stream_init(&_stream); mad_frame_init(&_frame); mad_synth_init(&_synth); _duration = duration; _posInFrame = 0; _bufferSize = size ? size : (128 * 1024); // Default buffer size is 128K _isStereo = false; _curChannel = 0; _file = file; _ptr = (byte *)malloc(_bufferSize + MAD_BUFFER_GUARD); init(); // If a size is specified, we do not perform any further read operations if (size) { _file = 0; } } MP3InputStream::~MP3InputStream() { mad_synth_finish(&_synth); mad_frame_finish(&_frame); mad_stream_finish(&_stream); free(_ptr); } bool MP3InputStream::init() { // TODO // Read in the first chunk of the MP3 file _size = _file->read(_ptr, _bufferSize); if (_size <= 0) { warning("MP3InputStream: Failed to read MP3 data"); return false; } // Feed the data we just read into the stream decoder mad_stream_buffer(&_stream, _ptr, _size); // Read in initial data refill(true); // Check the header, determine if this is a stereo stream int num; switch(_frame.header.mode) { case MAD_MODE_SINGLE_CHANNEL: case MAD_MODE_DUAL_CHANNEL: case MAD_MODE_JOINT_STEREO: case MAD_MODE_STEREO: num = MAD_NCHANNELS(&_frame.header); assert(num == 1 || num == 2); _isStereo = (num == 2); break; default: warning("MP3InputStream: Cannot determine number of channels"); return false; } return true; } void MP3InputStream::refill(bool first) { // Read the next frame (may have to retry several times, e.g. // to skip over ID3 information). while (mad_frame_decode(&_frame, &_stream)) { if (_stream.error == MAD_ERROR_BUFLEN) { int offset; if (!_file) _size = -1; // Give up immediately if we are at the EOF already if (_size <= 0) return; if (!_stream.next_frame) { offset = 0; memset(_ptr, 0, _bufferSize + MAD_BUFFER_GUARD); } else { offset = _stream.bufend - _stream.next_frame; memcpy(_ptr, _stream.next_frame, offset); } // Read in more data from the input file _size = _file->read(_ptr + offset, _bufferSize - offset); // Nothing read -> EOF -> bail out if (_size <= 0) { return; } _stream.error = (enum mad_error)0; // Feed the data we just read into the stream decoder mad_stream_buffer(&_stream, _ptr, _size + offset); } else if (MAD_RECOVERABLE(_stream.error)) { // FIXME: should we do anything here? debug(6, "MP3InputStream: Recoverable error..."); } else { error("MP3InputStream: Unrecoverable error"); } } // Subtract the duration of this frame from the time left to play mad_timer_t frame_duration = _frame.header.duration; mad_timer_negate(&frame_duration); mad_timer_add(&_duration, frame_duration); if (!first && _file && mad_timer_compare(_duration, mad_timer_zero) <= 0) _size = -1; // Mark for EOF // Synthesise the frame into PCM samples and reset the buffer position mad_synth_frame(&_synth, &_frame); _posInFrame = 0; } inline bool MP3InputStream::eosIntern() const { return (_size < 0 || _posInFrame >= _synth.pcm.length); } static inline int scale_sample(mad_fixed_t sample) { // round sample += (1L << (MAD_F_FRACBITS - 16)); // clip if (sample > MAD_F_ONE - 1) sample = MAD_F_ONE - 1; else if (sample < -MAD_F_ONE) sample = -MAD_F_ONE; // quantize and scale to not saturate when mixing a lot of channels return sample >> (MAD_F_FRACBITS + 1 - 16); } inline int16 MP3InputStream::read() { assert(!eosIntern()); int16 sample; if (_isStereo) { sample = (int16)scale_sample(_synth.pcm.samples[_curChannel][_posInFrame]); if (_curChannel == 0) { _curChannel = 1; } else { _posInFrame++; _curChannel = 0; } } else { sample = (int16)scale_sample(_synth.pcm.samples[0][_posInFrame]); _posInFrame++; } if (_posInFrame >= _synth.pcm.length) { refill(); } return sample; } int MP3InputStream::readBuffer(int16 *buffer, const int numSamples) { int samples = 0; assert(_curChannel == 0); // Paranoia check while (samples < numSamples && !eosIntern()) { const int len = MIN(numSamples, samples + (int)(_synth.pcm.length - _posInFrame) * (_isStereo ? 2 : 1)); while (samples < len) { *buffer++ = (int16)scale_sample(_synth.pcm.samples[0][_posInFrame]); samples++; if (_isStereo) { *buffer++ = (int16)scale_sample(_synth.pcm.samples[1][_posInFrame]); samples++; } _posInFrame++; } if (_posInFrame >= _synth.pcm.length) { refill(); } } return samples; } MusicStream *makeMP3Stream(File *file, mad_timer_t duration, uint size) { return new MP3InputStream(file, duration, size); } #endif #pragma mark - #pragma mark --- Ogg Vorbis stream --- #pragma mark - #ifdef USE_VORBIS class VorbisInputStream : public MusicStream { OggVorbis_File *_ov_file; int _end_pos; int _numChannels; int16 _buffer[4096]; const int16 *_bufferEnd; const int16 *_pos; void refill(); inline bool eosIntern() const; public: VorbisInputStream(OggVorbis_File *file, int duration); int readBuffer(int16 *buffer, const int numSamples); int16 read(); bool eos() const { return eosIntern(); } bool isStereo() const { return _numChannels >= 2; } int getRate() const { return ov_info(_ov_file, -1)->rate; } }; #ifdef CHUNKSIZE #define VORBIS_TREMOR #endif VorbisInputStream::VorbisInputStream(OggVorbis_File *file, int duration) : _ov_file(file), _bufferEnd(_buffer + ARRAYSIZE(_buffer)) { // Check the header, determine if this is a stereo stream _numChannels = ov_info(_ov_file, -1)->channels; // Determine the end position if (duration) _end_pos = ov_pcm_tell(_ov_file) + duration; else _end_pos = ov_pcm_total(_ov_file, -1); // Read in initial data refill(); } inline int16 VorbisInputStream::read() { assert(!eosIntern()); int16 sample = *_pos++; if (_pos >= _bufferEnd) { refill(); } return sample; } inline bool VorbisInputStream::eosIntern() const { return _pos >= _bufferEnd; } int VorbisInputStream::readBuffer(int16 *buffer, const int numSamples) { int samples = 0; while (samples < numSamples && !eosIntern()) { const int len = MIN(numSamples, samples + (int)(_bufferEnd - _pos)); memcpy(buffer, _pos, len * 2); buffer += len; _pos += len; samples += len; if (_pos >= _bufferEnd) { refill(); } } return samples; } void VorbisInputStream::refill() { // Read the samples uint len_left = sizeof(_buffer); char *read_pos = (char *)_buffer; while (len_left > 0 && _end_pos > ov_pcm_tell(_ov_file)) { long result = ov_read(_ov_file, read_pos, len_left, #ifndef VORBIS_TREMOR #ifdef SCUMM_BIG_ENDIAN 1, #else 0, #endif 2, // 16 bit 1, // signed #endif NULL); if (result == OV_HOLE) { // Possibly recoverable, just warn about it warning("Corrupted data in Vorbis file"); } else if (result <= 0) { if (result < 0) debug(1, "Decode error %d in Vorbis file", result); // Don't delete it yet, that causes problems in // the CD player emulation code. memset(read_pos, 0, len_left); break; } else { len_left -= result; read_pos += result; } } _pos = _buffer; _bufferEnd = (int16 *)read_pos; } MusicStream *makeVorbisStream(OggVorbis_File *file, int duration) { return new VorbisInputStream(file, duration); } #endif #pragma mark - #pragma mark --- Input stream factories --- #pragma mark - template static AudioInputStream *makeLinearInputStream(const byte *ptr, uint32 len, bool is16Bit, bool isUnsigned, bool isLE, uint loopOffset, uint loopLen) { if (isUnsigned) { if (is16Bit) { if (isLE) return new LinearMemoryStream(ptr, len, loopOffset, loopLen); else return new LinearMemoryStream(ptr, len, loopOffset, loopLen); } else return new LinearMemoryStream(ptr, len, loopOffset, loopLen); } else { if (is16Bit) { if (isLE) return new LinearMemoryStream(ptr, len, loopOffset, loopLen); else return new LinearMemoryStream(ptr, len, loopOffset, loopLen); } else return new LinearMemoryStream(ptr, len, loopOffset, loopLen); } } template static WrappedAudioInputStream *makeWrappedInputStream(uint32 len, bool is16Bit, bool isUnsigned) { if (isUnsigned) { if (is16Bit) return new WrappedMemoryStream(len); else return new WrappedMemoryStream(len); } else { if (is16Bit) return new WrappedMemoryStream(len); else return new WrappedMemoryStream(len); } } AudioInputStream *makeLinearInputStream(byte _flags, const byte *ptr, uint32 len, uint loopOffset, uint loopLen) { const bool is16Bit = (_flags & SoundMixer::FLAG_16BITS) != 0; const bool isUnsigned = (_flags & SoundMixer::FLAG_UNSIGNED) != 0; const bool isLE = (_flags & SoundMixer::FLAG_LITTLE_ENDIAN) != 0; if (_flags & SoundMixer::FLAG_STEREO) { return makeLinearInputStream(ptr, len, is16Bit, isUnsigned, isLE, loopOffset, loopLen); } else { return makeLinearInputStream(ptr, len, is16Bit, isUnsigned, isLE, loopOffset, loopLen); } } WrappedAudioInputStream *makeWrappedInputStream(byte _flags, uint32 len) { const bool is16Bit = (_flags & SoundMixer::FLAG_16BITS) != 0; const bool isUnsigned = (_flags & SoundMixer::FLAG_UNSIGNED) != 0; if (_flags & SoundMixer::FLAG_STEREO) { return makeWrappedInputStream(len, is16Bit, isUnsigned); } else { return makeWrappedInputStream(len, is16Bit, isUnsigned); } }