mirror of
https://github.com/libretro/scummvm.git
synced 2024-12-24 02:36:27 +00:00
9b2d4f92aa
svn-id: r9313
474 lines
11 KiB
C++
474 lines
11 KiB
C++
/* ScummVM - Scumm Interpreter
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* Copyright (C) 2001-2003 The ScummVM project
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* as published by the Free Software Foundation; either version 2
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* of the License, or (at your option) any later version.
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
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*
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* $Header$
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*
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*/
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#include "stdafx.h"
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#include "audiostream.h"
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#include "mixer.h"
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#include "common/engine.h"
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#include "common/file.h"
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#include "common/util.h"
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template<bool is16Bit, bool isUnsigned>
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static inline int16 readSample(const byte *ptr) {
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uint16 sample = is16Bit ? READ_BE_UINT16(ptr) : (*ptr << 8);
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if (isUnsigned)
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sample ^= 0x8000;
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return (int16)sample;
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}
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#pragma mark -
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#pragma mark --- LinearMemoryStream ---
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#pragma mark -
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template<bool stereo, bool is16Bit, bool isUnsigned>
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class LinearMemoryStream : public AudioInputStream {
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protected:
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const byte *_ptr;
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const byte *_end;
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public:
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LinearMemoryStream(const byte *ptr, uint len)
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: _ptr(ptr), _end(ptr+len) {
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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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return val;
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}
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bool eof() const {
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return _end <= _ptr;
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}
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bool isStereo() const {
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return stereo;
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}
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};
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#pragma mark -
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#pragma mark --- WrappedMemoryStream ---
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#pragma mark -
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// Wrapped memory stream, to be used by the ChannelStream class (and possibly others?)
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template<bool stereo, bool is16Bit, bool isUnsigned>
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class WrappedMemoryStream : public WrappedAudioInputStream {
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protected:
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byte *_bufferStart;
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byte *_bufferEnd;
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byte *_pos;
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byte *_end;
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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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void append(const byte *data, uint32 len);
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};
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template<bool stereo, bool is16Bit, bool isUnsigned>
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WrappedMemoryStream<stereo, is16Bit, isUnsigned>::WrappedMemoryStream(uint bufferSize) {
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if (stereo) // Stereo requires an even sized buffer
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assert(bufferSize % 2 == 0);
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_bufferStart = (byte *)malloc(bufferSize);
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_pos = _end = _bufferStart;
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_bufferEnd = _bufferStart + bufferSize;
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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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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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// Wrap around?
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if (_pos >= _bufferEnd)
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_pos = _pos - (_bufferEnd - _bufferStart);
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return val;
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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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}
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template<bool stereo, bool is16Bit, bool isUnsigned>
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void WrappedMemoryStream<stereo, is16Bit, isUnsigned>::append(const byte *data, uint32 len) {
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if (_end + len > _bufferEnd) {
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// Wrap-around case
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uint32 size_to_end_of_buffer = _bufferEnd - _end;
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len -= size_to_end_of_buffer;
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if ((_end < _pos) || (_bufferStart + len >= _pos)) {
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debug(2, "WrappedMemoryStream: buffer overflow (A)");
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return;
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}
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memcpy(_end, data, size_to_end_of_buffer);
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memcpy(_bufferStart, data + size_to_end_of_buffer, len);
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_end = _bufferStart + len;
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} else {
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if ((_end < _pos) && (_end + len >= _pos)) {
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debug(2, "WrappedMemoryStream: buffer overflow (B)");
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return;
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}
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memcpy(_end, data, len);
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_end += len;
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}
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}
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#pragma mark -
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#pragma mark --- MP3 (MAD) stream ---
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#pragma mark -
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#ifdef USE_MAD
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#define MP3_BUFFER_SIZE 131072
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/**
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* Playback the MP3 data in the given file for the specified duration.
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*
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* @param file file containing the MP3 data
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* @param duration playback duration in frames (1/75th of a second), 0 means playback until EOF
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*/
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MP3InputStream::MP3InputStream(File *file, mad_timer_t duration) {
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// duration == 0 means: play everything till end of file
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_isStereo = false;
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_curChannel = 0;
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_file = file;
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_rate = 0;
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_posInFrame = 0;
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_duration = duration;
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mad_stream_init(&_stream);
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mad_frame_init(&_frame);
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mad_synth_init(&_synth);
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_ptr = (byte *)malloc(MP3_BUFFER_SIZE + MAD_BUFFER_GUARD);
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_initialized = init();
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}
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MP3InputStream::~MP3InputStream() {
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mad_synth_finish(&_synth);
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mad_frame_finish(&_frame);
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mad_stream_finish(&_stream);
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free(_ptr);
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}
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bool MP3InputStream::init() {
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// TODO
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// Read in the first chunk of the MP3 file
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_size = _file->read(_ptr, MP3_BUFFER_SIZE);
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if (_size <= 0) {
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warning("MP3InputStream: Failed to read MP3 data");
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return false;
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}
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// Feed the data we just read into the stream decoder
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mad_stream_buffer(&_stream, _ptr, _size);
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// Read in initial data
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refill();
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// Check the header, determine if this is a stereo stream
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int num;
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switch(_frame.header.mode)
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{
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case MAD_MODE_SINGLE_CHANNEL:
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case MAD_MODE_DUAL_CHANNEL:
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case MAD_MODE_JOINT_STEREO:
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case MAD_MODE_STEREO:
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num = MAD_NCHANNELS(&_frame.header);
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assert(num == 1 || num == 2);
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_isStereo = (num == 2);
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break;
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default:
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warning("MP3InputStream: Cannot determine number of channels");
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return false;
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}
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// Determine the sample rate
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_rate = _frame.header.samplerate;
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return true;
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}
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void MP3InputStream::refill() {
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// Read the next frame (may have to retry several times, e.g.
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// to skip over ID3 information).
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while (mad_frame_decode(&_frame, &_stream)) {
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if (_stream.error == MAD_ERROR_BUFLEN) {
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int offset;
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// Give up immediately if we are at the EOF already
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if (_size <= 0)
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return;
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if (!_stream.next_frame) {
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offset = 0;
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memset(_ptr, 0, MP3_BUFFER_SIZE + MAD_BUFFER_GUARD);
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} else {
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offset = _stream.bufend - _stream.next_frame;
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memcpy(_ptr, _stream.next_frame, offset);
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}
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// Read in more data from the input file
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_size = _file->read(_ptr + offset, MP3_BUFFER_SIZE - offset);
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// Nothing read -> EOF -> bail out
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if (_size <= 0) {
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return;
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}
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_stream.error = (enum mad_error)0;
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// Feed the data we just read into the stream decoder
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mad_stream_buffer(&_stream, _ptr, _size + offset);
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} else if (MAD_RECOVERABLE(_stream.error)) {
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// FIXME: should we do anything here?
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warning("MP3InputStream: Recoverable error...");
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} else {
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error("MP3InputStream: Unrecoverable error");
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}
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}
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// Subtract the duration of this frame from the time left to play
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mad_timer_t frame_duration = _frame.header.duration;
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mad_timer_negate(&frame_duration);
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mad_timer_add(&_duration, _frame.header.duration);
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// Synthesise the frame into PCM samples and reset the buffer position
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mad_synth_frame(&_synth, &_frame);
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_posInFrame = 0;
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}
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bool MP3InputStream::eof() const {
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// Time over -> input steam ends
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if (mad_timer_compare(_duration, mad_timer_zero) <= 0)
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return true;
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// Data left in the PCM buffer -> we are not yet done!
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if (_posInFrame < _synth.pcm.length)
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return false;
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// EOF of the input file, we are done
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if (_size < 0)
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return true;
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// Otherwise, we are still good to go
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return false;
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}
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static inline int scale_sample(mad_fixed_t sample) {
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// round
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sample += (1L << (MAD_F_FRACBITS - 16));
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// clip
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if (sample > MAD_F_ONE - 1)
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sample = MAD_F_ONE - 1;
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else if (sample < -MAD_F_ONE)
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sample = -MAD_F_ONE;
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// quantize and scale to not saturate when mixing a lot of channels
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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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if (_posInFrame >= _synth.pcm.length) {
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refill();
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if (_size < 0) // EOF
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return 0;
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}
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int16 sample;
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if (_isStereo) {
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sample = (int16)scale_sample(_synth.pcm.samples[_curChannel][_posInFrame]);
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if (_curChannel == 0) {
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_curChannel = 1;
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} else {
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_posInFrame++;
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_curChannel = 0;
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}
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} else {
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sample = (int16)scale_sample(_synth.pcm.samples[0][_posInFrame]);
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_posInFrame++;
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}
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return sample;
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}
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#endif
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#pragma mark -
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#pragma mark --- Ogg Vorbis stream ---
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#pragma mark -
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#ifdef USE_VORBIS
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#ifdef CHUNKSIZE
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#define VORBIS_TREMOR
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#endif
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VorbisInputStream::VorbisInputStream(OggVorbis_File *file, int duration)
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: _ov_file(file) {
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_pos = _buffer + ARRAYSIZE(_buffer);
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_numChannels = ov_info(_ov_file, -1)->channels;
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if (duration)
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_end_pos = ov_pcm_tell(_ov_file) + duration;
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else
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_end_pos = ov_pcm_total(_ov_file, -1);
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_eof_flag = false;
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}
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int16 VorbisInputStream::read() {
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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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if (_eof_flag)
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return true;
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if (_pos < _buffer + ARRAYSIZE(_buffer))
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return false;
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return (_end_pos <= ov_pcm_tell(_ov_file));
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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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char *read_pos = (char *)_buffer;
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while (len_left > 0) {
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long result = ov_read(_ov_file, read_pos, len_left,
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#ifndef VORBIS_TREMOR
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#ifdef SCUMM_BIG_ENDIAN
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1,
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#else
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0,
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#endif
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2, // 16 bit
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1, // signed
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#endif
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NULL);
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if (result == 0) {
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_eof_flag = true;
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memset(read_pos, 0, len_left);
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break;
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} else if (result == OV_HOLE) {
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// Possibly recoverable, just warn about it
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warning("Corrupted data in Vorbis file");
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} else if (result < 0) {
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debug(1, "Decode error %d in Vorbis file", result);
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// Don't delete it yet, that causes problems in
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// the CD player emulation code.
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_eof_flag = true;
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memset(read_pos, 0, len_left);
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break;
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} else {
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len_left -= result;
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read_pos += result;
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}
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}
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_pos = _buffer;
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}
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#endif
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#pragma mark -
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#pragma mark --- Input stream factories ---
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#pragma mark -
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template<bool stereo>
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static AudioInputStream *makeLinearInputStream(const byte *ptr, uint32 len, bool is16Bit, bool isUnsigned) {
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if (isUnsigned) {
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if (is16Bit)
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return new LinearMemoryStream<stereo, true, true>(ptr, len);
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else
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return new LinearMemoryStream<stereo, false, true>(ptr, len);
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} else {
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if (is16Bit)
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return new LinearMemoryStream<stereo, true, false>(ptr, len);
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else
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return new LinearMemoryStream<stereo, false, false>(ptr, len);
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}
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}
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template<bool stereo>
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static WrappedAudioInputStream *makeWrappedInputStream(uint32 len, bool is16Bit, bool isUnsigned) {
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if (isUnsigned) {
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if (is16Bit)
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return new WrappedMemoryStream<stereo, true, true>(len);
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else
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return new WrappedMemoryStream<stereo, false, true>(len);
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} else {
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if (is16Bit)
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return new WrappedMemoryStream<stereo, true, false>(len);
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else
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return new WrappedMemoryStream<stereo, false, false>(len);
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}
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}
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AudioInputStream *makeLinearInputStream(byte _flags, const byte *ptr, uint32 len) {
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const bool is16Bit = (_flags & SoundMixer::FLAG_16BITS) != 0;
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const bool isUnsigned = (_flags & SoundMixer::FLAG_UNSIGNED) != 0;
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if (_flags & SoundMixer::FLAG_STEREO) {
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return makeLinearInputStream<true>(ptr, len, is16Bit, isUnsigned);
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} else {
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return makeLinearInputStream<false>(ptr, len, is16Bit, isUnsigned);
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}
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}
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WrappedAudioInputStream *makeWrappedInputStream(byte _flags, uint32 len) {
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const bool is16Bit = (_flags & SoundMixer::FLAG_16BITS) != 0;
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const bool isUnsigned = (_flags & SoundMixer::FLAG_UNSIGNED) != 0;
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if (_flags & SoundMixer::FLAG_STEREO) {
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return makeWrappedInputStream<true>(len, is16Bit, isUnsigned);
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} else {
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return makeWrappedInputStream<false>(len, is16Bit, isUnsigned);
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}
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}
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