ScummVM

Revision: 25756
          http://scummvm.svn.sourceforge.net/scummvm/?rev=25756&view=rev
Author:   fingolfin
Date:     2007-02-20 14:18:48 -0800 (Tue, 20 Feb 2007)

Log Message:
-----------
Added looping to the MP3 streams (currently virtually untested, so watch out)

Modified Paths:
--------------
    scummvm/trunk/sound/mp3.cpp
    scummvm/trunk/sound/mp3.h

Modified: scummvm/trunk/sound/mp3.cpp
===================================================================
--- scummvm/trunk/sound/mp3.cpp	2007-02-20 21:53:22 UTC (rev 25755)
+++ scummvm/trunk/sound/mp3.cpp	2007-02-20 22:18:48 UTC (rev 25756)
@@ -47,12 +47,11 @@
 	mad_frame _frame;
 	mad_synth _synth;
 
-	mad_timer_t _startTime;	
-	mad_timer_t _endTime;
+	const mad_timer_t _startTime;	
+	const mad_timer_t _endTime;
 	mad_timer_t _totalTime;
 	
-	// TODO: For looping, we will have to use a SeekableReadStream here
-	Common::ReadStream *_inStream;
+	Common::SeekableReadStream *_inStream;
 	bool _disposeAfterUse;
 	
 	enum {
@@ -62,17 +61,16 @@
 	// This buffer contains a slab of input data
 	byte _buf[BUFFER_SIZE + MAD_BUFFER_GUARD];
 
-
-	uint32 _posInFrame;
-	int _curChannel;
-	
+	uint _numLoops;
+	uint _posInFrame;
 	bool _eos;
 	
 public:
-	MP3InputStream(Common::ReadStream *inStream,
+	MP3InputStream(Common::SeekableReadStream *inStream,
 	               bool dispose,
 	               mad_timer_t start = mad_timer_zero,
-	               mad_timer_t end = mad_timer_zero);
+	               mad_timer_t end = mad_timer_zero,
+	               uint numLoops = 1);
 	~MP3InputStream();
 	
 	bool init();
@@ -84,24 +82,24 @@
 	int getRate() const			{ return _frame.header.samplerate; }
 
 protected:
+	void rewind();
 	void decodeMP3Data();
 	bool readMP3Data();
 };
 
-MP3InputStream::MP3InputStream(Common::ReadStream *inStream, bool dispose, mad_timer_t start, mad_timer_t end) :
+MP3InputStream::MP3InputStream(Common::SeekableReadStream *inStream, bool dispose, mad_timer_t start, mad_timer_t end, uint numLoops) :
 	_inStream(inStream),
 	_disposeAfterUse(dispose),
 	_startTime(start),
 	_endTime(end),
 	_totalTime(mad_timer_zero),
+	_numLoops(numLoops),
+	_posInFrame(0),
 	_eos(false) {
 
 	// Make sure that either start < end, or end is zero (indicating "play until end")
 	assert(mad_timer_compare(_startTime, _endTime) < 0 || mad_timer_sign(_endTime) == 0);
 
-	_posInFrame = 0;
-	_curChannel = 0;
-
 	// The MAD_BUFFER_GUARD must always contain zeros (the reason
 	// for this is that the Layer III Huffman decoder of libMAD
 	// may read a few bytes beyond the end of the input buffer).
@@ -126,6 +124,24 @@
 		delete _inStream;
 }
 
+void MP3InputStream::rewind() {
+	// Start over again: reset the decoders, seek back to the start of the file, etc.
+
+	mad_synth_finish(&_synth);
+	mad_frame_finish(&_frame);
+	mad_stream_finish(&_stream);
+	
+	_inStream->seek(0, SEEK_SET);
+	_totalTime = mad_timer_zero;
+	_posInFrame = 0;
+	_eos = false;
+
+	// Reinit MAD
+	mad_stream_init(&_stream);
+	mad_frame_init(&_frame);
+	mad_synth_init(&_synth);
+}
+
 void MP3InputStream::decodeMP3Data() {
 	if (_eos)
 		return;
@@ -141,7 +157,7 @@
 			}
 		}
 
-		while (true) {
+		while (!_eos) {
 			_stream.error = MAD_ERROR_NONE;
 
 			// Decode the next header. Note: mad_frame_decode would do this for us, too.
@@ -191,6 +207,12 @@
 			_posInFrame = 0;
 			break;
 		}
+	
+		if (_eos) {
+			// If looping is enabled, try again
+			if (_numLoops == 0 || --_numLoops > 0)
+				rewind();
+		}
 
 	} while (_stream.error == MAD_ERROR_BUFLEN);
 	
@@ -244,7 +266,6 @@
 
 int MP3InputStream::readBuffer(int16 *buffer, const int numSamples) {
 	int samples = 0;
-	assert(_curChannel == 0);	// Paranoia check
 	// Keep going as long as we have input available
 	while (samples < numSamples && !_eos) {
 		const int len = MIN(numSamples, samples + (int)(_synth.pcm.length - _posInFrame) * MAD_NCHANNELS(&_frame.header));
@@ -283,7 +304,32 @@
 	return new MP3InputStream(stream, true);
 }
 
+AudioStream *makeMP3Stream(
+	Common::SeekableReadStream *stream,
+	bool disposeAfterUse,
+	uint32 startTime,
+	uint32 duration,
+	uint numLoops) {
 
+	mad_timer_t start;
+	mad_timer_t end;
+
+	// Both startTime and duration are given in milliseconds.
+	// Calculate the appropriate mad_timer_t values from them.
+	mad_timer_set(&start, startTime / 1000, startTime % 1000, 1000);
+	if (duration == 0) {
+		end = mad_timer_zero;
+	} else {
+		int endTime = startTime + duration;
+		mad_timer_set(&end, endTime / 1000, endTime % 1000, 1000);
+	}
+
+	MP3InputStream *mp3Stream = new MP3InputStream(stream, disposeAfterUse, start, end, numLoops);
+	
+	return mp3Stream;
+}
+
+
 #pragma mark -
 #pragma mark --- MP3 Audio CD emulation ---
 #pragma mark -

Modified: scummvm/trunk/sound/mp3.h
===================================================================
--- scummvm/trunk/sound/mp3.h	2007-02-20 21:53:22 UTC (rev 25755)
+++ scummvm/trunk/sound/mp3.h	2007-02-20 22:18:48 UTC (rev 25756)
@@ -30,6 +30,7 @@
 
 namespace Common {
 	class File;
+	class SeekableReadStream;
 }
 
 namespace Audio {
@@ -47,6 +48,27 @@
  */
 AudioStream *makeMP3Stream(Common::File *file, uint32 size);
 
+
+/**
+ * Create a new AudioStream from the MP3 data in the given stream.
+ * Allows for looping (which is why we require a SeekableReadStream),
+ * and specifying only a portion of the data to be played, based 
+ * on time offsets.
+ *
+ * @param stream			the SeekableReadStream from which to read the MP3 data
+ * @param disposeAfterUse	whether to delete the stream after use
+ * @param startTime			the (optional) time offset in milliseconds from which to start playback 
+ * @param duration			the (optional) time in milliseconds specifying how long to play
+ * @param numLoops			how often the data shall be looped (0 = infinite)
+ * @return	a new AudioStream, or NULL, if an error occured
+ */
+AudioStream *makeMP3Stream(
+	Common::SeekableReadStream *stream,
+	bool disposeAfterUse,
+	uint32 startTime = 0,
+	uint32 duration = 0,
+	uint numLoops = 1);
+
 } // End of namespace Audio
 
 #endif // #ifdef USE_MAD


This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.

[Scummvm-cvs-logs] SF.net SVN: scummvm: [25757] scummvm/trunk/sound From: <fingolfin@us...> - 2007-02-20 23:40 Revision: 25757 http://scummvm.svn.sourceforge.net/scummvm/?rev=25757&view=rev Author: fingolfin Date: 2007-02-20 15:40:46 -0800 (Tue, 20 Feb 2007) Log Message: ----------- Revamed the Ogg Vorbis & FLAC DigitalTrackInfo subclasses to work similar to the MP3 one (i.e. only open the data file when about to play) Modified Paths: -------------- scummvm/trunk/sound/flac.cpp scummvm/trunk/sound/mp3.cpp scummvm/trunk/sound/vorbis.cpp Modified: scummvm/trunk/sound/flac.cpp =================================================================== --- scummvm/trunk/sound/flac.cpp 2007-02-20 22:18:48 UTC (rev 25756) +++ scummvm/trunk/sound/flac.cpp 2007-02-20 23:40:46 UTC (rev 25757) @@ -874,41 +874,57 @@ class FlacTrackInfo : public DigitalTrackInfo { private: - File *_file; - FlacInputStream *_firstStream; // avoid having to open the Stream twice the first time + Common::String _filename; + bool _errorFlag; public: - FlacTrackInfo(File *file); - ~FlacTrackInfo(); - bool error() { return _file == NULL; } + FlacTrackInfo(const char *filename); + bool error() { return _errorFlag; } void play(Audio::Mixer *mixer, Audio::SoundHandle *handle, int startFrame, int duration); }; -FlacTrackInfo::FlacTrackInfo(File *file) : _file(NULL), _firstStream(NULL) { - FlacInputStream *tempStream = new FlacInputStream(file); - /* first time the file will be tested, but not used */ - if (tempStream->init()) { - _firstStream = tempStream; - _file = file; - } else - delete tempStream; +FlacTrackInfo::FlacTrackInfo(const char *filename) : + _filename(filename), + _errorFlag(false) { + + // Try to open the file + Common::File file; + if (!file.open(_filename)) { + _errorFlag = true; + return; + } + + // Next, try to create a FlacInputStream from it + FlacInputStream *tempStream = new FlacInputStream(&file); + + // If initialising the stream fails, we set the error flag + if (!tempStream || !tempStream->init()) + _errorFlag = true; + + delete tempStream; } void FlacTrackInfo::play(Audio::Mixer *mixer, Audio::SoundHandle *handle, int startFrame, int duration) { + assert(!_errorFlag); + if (error()) { debug(1, "FlacTrackInfo::play: invalid state, method should not been called"); } - FlacInputStream *flac; - - if (_firstStream != NULL) { - flac = _firstStream; - _firstStream = NULL; - } else { - flac = new FlacInputStream(_file); - flac->init(); + // Open the file + Common::File *file = new Common::File(); + if (!file || !file->open(_filename)) { + warning("FlacTrackInfo::play: failed to open '%s'", _filename.c_str()); + delete file; + return; } + // Create an AudioStream from the file + FlacInputStream *flac = new FlacInputStream(file); + flac->init(); + file->decRef(); + + // Seek to the correct start position and start playback if (flac->isStreamDecoderReady()) { const FLAC__StreamMetadata_StreamInfo &info = flac->getStreamInfo(); if (duration) @@ -921,38 +937,30 @@ return; } // startSample is beyond the existing Samples - debug(1, "FlacTrackInfo: Audiostream %s could not seek to frame %d (ca %d secs)", _file->name(), startFrame, startFrame/75); + debug(1, "FlacTrackInfo: Audiostream %s could not seek to frame %d (ca %d secs)", _filename.c_str(), startFrame, startFrame/75); flac->finish(); } delete flac; } -FlacTrackInfo::~FlacTrackInfo() { - delete _firstStream; - delete _file; -} - DigitalTrackInfo* getFlacTrack(int track) { assert(track >= 1); char trackName[4][32]; - File *file = new File(); sprintf(trackName[0], "track%d.flac", track); sprintf(trackName[1], "track%02d.flac", track); sprintf(trackName[2], "track%d.fla", track); sprintf(trackName[3], "track%02d.fla", track); - for (int i = 0; i < 4; ++i) { - if (file->open(trackName[i])) { - FlacTrackInfo *trackInfo = new FlacTrackInfo(file); + if (Common::File::exists(trackName[i])) { + FlacTrackInfo *trackInfo = new FlacTrackInfo(trackName[i]); if (!trackInfo->error()) return trackInfo; delete trackInfo; } } - delete file; return NULL; } Modified: scummvm/trunk/sound/mp3.cpp =================================================================== --- scummvm/trunk/sound/mp3.cpp 2007-02-20 22:18:48 UTC (rev 25756) +++ scummvm/trunk/sound/mp3.cpp 2007-02-20 23:40:46 UTC (rev 25757) @@ -167,7 +167,7 @@ if (_stream.error == MAD_ERROR_BUFLEN) { break; // Read more data } else if (MAD_RECOVERABLE(_stream.error)) { - debug(1, "MP3InputStream: Recoverable error in mad_header_decode (%s)", mad_stream_errorstr(&_stream)); + debug(6, "MP3InputStream: Recoverable error in mad_header_decode (%s)", mad_stream_errorstr(&_stream)); continue; } else { warning("MP3InputStream: Unrecoverable error in mad_header_decode (%s)", mad_stream_errorstr(&_stream)); @@ -193,8 +193,10 @@ if (_stream.error == MAD_ERROR_BUFLEN) { break; // Read more data } else if (MAD_RECOVERABLE(_stream.error)) { - // FIXME: should we do anything here? - debug(1, "MP3InputStream: Recoverable error in mad_frame_decode (%s)", mad_stream_errorstr(&_stream)); + // Note: we will occasionally see MAD_ERROR_BADDATAPTR errors here. + // These are normal and expected (caused by our frame skipping (i.e. "seeking") + // code above). + debug(6, "MP3InputStream: Recoverable error in mad_frame_decode (%s)", mad_stream_errorstr(&_stream)); continue; } else { warning("MP3InputStream: Unrecoverable error in mad_frame_decode (%s)", mad_stream_errorstr(&_stream)); @@ -351,27 +353,27 @@ _filename(filename), _errorFlag(false) { + // Try to open the file Common::File file; - - // Try to open the file if (!file.open(_filename)) { _errorFlag = true; return; } - // Next, try to create a MP3InputStream from it + // Next, try to create an MP3InputStream from it + MP3InputStream *tempStream = new MP3InputStream(&file, false); - MP3InputStream *mp3Stream = new MP3InputStream(&file, false); - // If we see EOS here then that means that not (enough) valid input // data was given. - _errorFlag = mp3Stream->endOfData(); + _errorFlag = tempStream->endOfData(); // Clean up again - delete mp3Stream; + delete tempStream; } void MP3TrackInfo::play(Audio::Mixer *mixer, Audio::SoundHandle *handle, int startFrame, int duration) { + assert(!_errorFlag); + mad_timer_t start; mad_timer_t end; @@ -389,12 +391,14 @@ Common::File *file = new Common::File(); if (!file || !file->open(_filename)) { warning("MP3TrackInfo::play: failed to open '%s'", _filename.c_str()); + delete file; return; } - // Play it + // ... create an AudioStream ... MP3InputStream *input = new MP3InputStream(file, true, start, end); + // ... and play it mixer->playInputStream(Audio::Mixer::kMusicSoundType, handle, input); } Modified: scummvm/trunk/sound/vorbis.cpp =================================================================== --- scummvm/trunk/sound/vorbis.cpp 2007-02-20 22:18:48 UTC (rev 25756) +++ scummvm/trunk/sound/vorbis.cpp 2007-02-20 23:40:46 UTC (rev 25757) @@ -46,8 +46,6 @@ namespace Audio { -static AudioStream *makeVorbisStream(OggVorbis_File *file, int duration); - // These are wrapper functions to allow using a File object to // provide data to the OggVorbis_File object. @@ -129,8 +127,9 @@ static int close_wrap(void *datasource) { file_info *f = (file_info *) datasource; - f->file->decRef(); + delete f->file; delete f; + return 0; } @@ -146,123 +145,6 @@ #pragma mark - -#pragma mark --- Ogg Vorbis Audio CD emulation --- -#pragma mark - - -class VorbisTrackInfo : public DigitalTrackInfo { -private: - File *_file; - OggVorbis_File _ov_file; - bool _error_flag; - -public: - VorbisTrackInfo(File *file); - ~VorbisTrackInfo(); - bool openTrack(); - bool error() { return _error_flag; } - void play(Audio::Mixer *mixer, Audio::SoundHandle *handle, int startFrame, int duration); -}; - - -VorbisTrackInfo::VorbisTrackInfo(File *file) { -//debug(5, "" __FILE__ ":%i", __LINE__); - - _file = file; - if (openTrack()) { - warning("Invalid file format"); - _error_flag = true; - _file = 0; - } else { - _error_flag = false; - _file->incRef(); - ov_clear(&_ov_file); - } -} - -bool VorbisTrackInfo::openTrack() { -//debug(5, "" __FILE__ ":%i", __LINE__); - assert(_file); - - file_info *f = new file_info; - -#if defined(__SYMBIAN32__) - // Symbian can't share filehandles between different threads. - // So create a new file and seek that to the other filehandles position - f->file = new File; - f->file->open(_file->name()); - f->file->seek(_file->pos()); -#else - f->file = _file; -#endif - - f->start = 0; - f->len = _file->size(); - f->curr_pos = 0; - _file->seek(0); - - bool err = (ov_open_callbacks((void *) f, &_ov_file, NULL, 0, g_File_wrap) < 0); - if (err) { -#ifdef __SYMBIAN32__ - delete f->file; -#endif - delete f; - } else { -#ifndef __SYMBIAN32__ - _file->incRef(); -#endif - } - - return err; -} - -VorbisTrackInfo::~VorbisTrackInfo() { -//debug(5, "" __FILE__ ":%i", __LINE__); - if (! _error_flag) { - ov_clear(&_ov_file); - _file->decRef(); - } -} - -void VorbisTrackInfo::play(Audio::Mixer *mixer, Audio::SoundHandle *handle, int startFrame, int duration) { -//debug(5, "" __FILE__ ":%i", __LINE__); - bool err = openTrack(); err=err;//satisfy unused variable - assert(!err); - -#ifdef USE_TREMOR // In Tremor, the ov_time_seek() and ov_time_seek_page() calls take seeking positions in milliseconds as 64 bit integers, rather than in seconds as doubles as in Vorbisfile. -#if defined(__SYMBIAN32__) && defined(__GCC32__) // SumthinWicked says: fixing "relocation truncated to fit: ARM_26 __fixdfdi" during linking on GCC, see portdefs.h - ov_time_seek(&_ov_file, (ogg_int64_t)scumm_fixdfdi(startFrame / 75.0 * 1000)); -#else - ov_time_seek(&_ov_file, (ogg_int64_t)(startFrame / 75.0 * 1000)); -#endif -#else - ov_time_seek(&_ov_file, startFrame / 75.0); -#endif - - AudioStream *input = makeVorbisStream(&_ov_file, duration * ov_info(&_ov_file, -1)->rate / 75); - mixer->playInputStream(Audio::Mixer::kMusicSoundType, handle, input); -} - -DigitalTrackInfo *getVorbisTrack(int track) { - char trackName[2][32]; - File *file = new File(); - - sprintf(trackName[0], "track%d.ogg", track); - sprintf(trackName[1], "track%02d.ogg", track); - - for (int i = 0; i < 2; ++i) { - if (file->open(trackName[i])) { - VorbisTrackInfo *trackInfo = new VorbisTrackInfo(file); - file->decRef(); - if (!trackInfo->error()) - return trackInfo; - delete trackInfo; - } - } - delete file; - return NULL; -} - -#pragma mark - #pragma mark --- Ogg Vorbis stream --- #pragma mark - @@ -376,11 +258,6 @@ _bufferEnd = (int16 *)read_pos; } -static AudioStream *makeVorbisStream(OggVorbis_File *file, int duration) { -//debug(5, "" __FILE__ ":%i", __LINE__); - return new VorbisInputStream(file, duration, false); -} - AudioStream *makeVorbisStream(File *file, uint32 size) { //debug(5, "" __FILE__ ":%i", __LINE__); OggVorbis_File *ov_file = new OggVorbis_File; @@ -413,6 +290,102 @@ } +#pragma mark - +#pragma mark --- Ogg Vorbis Audio CD emulation --- +#pragma mark - + +class VorbisTrackInfo : public DigitalTrackInfo { +private: + Common::String _filename; + bool _errorFlag; + +public: + VorbisTrackInfo(const char *filename); + bool openTrack(OggVorbis_File *ovFile); + bool error() { return _errorFlag; } + void play(Audio::Mixer *mixer, Audio::SoundHandle *handle, int startFrame, int duration); +}; + + +VorbisTrackInfo::VorbisTrackInfo(const char *filename) : + _filename(filename), + _errorFlag(false) { + + OggVorbis_File ov_file; + _errorFlag = openTrack(&ov_file); + + if (!_errorFlag) + ov_clear(&ov_file); +} + +bool VorbisTrackInfo::openTrack(OggVorbis_File *ovFile) { + bool err; + + file_info *f = new file_info; + assert(f); + f->file = new File; + assert(f->file); + + err = !f->file->open(_filename); + + if (!err) { + f->start = 0; + f->len = f->file->size(); + f->curr_pos = 0; + + err = (ov_open_callbacks((void *) f, ovFile, NULL, 0, g_File_wrap) < 0); + } + + if (err) { + delete f->file; + delete f; + } + + return err; +} + +void VorbisTrackInfo::play(Audio::Mixer *mixer, Audio::SoundHandle *handle, int startFrame, int duration) { + OggVorbis_File *ovFile = new OggVorbis_File; + bool err = openTrack(ovFile); + assert(!err); + +#ifdef USE_TREMOR + // In Tremor, the ov_time_seek() and ov_time_seek_page() calls take seeking + // positions in milliseconds as 64 bit integers, rather than in seconds as + // doubles as in Vorbisfile. +#if defined(__SYMBIAN32__) && defined(__GCC32__) + // SumthinWicked says: fixing "relocation truncated to fit: ARM_26 __fixdfdi" during linking on GCC, see portdefs.h + ov_time_seek(ovFile, (ogg_int64_t)scumm_fixdfdi(startFrame / 75.0 * 1000)); +#else + ov_time_seek(ovFile, (ogg_int64_t)(startFrame / 75.0 * 1000)); +#endif +#else + ov_time_seek(ovFile, startFrame / 75.0); +#endif + + AudioStream *input = new VorbisInputStream(ovFile, duration * ov_info(ovFile, -1)->rate / 75, true); + mixer->playInputStream(Audio::Mixer::kMusicSoundType, handle, input); +} + +DigitalTrackInfo *getVorbisTrack(int track) { + char trackName[2][32]; + + sprintf(trackName[0], "track%d.ogg", track); + sprintf(trackName[1], "track%02d.ogg", track); + + for (int i = 0; i < 2; ++i) { + if (Common::File::exists(trackName[i])) { + VorbisTrackInfo *trackInfo = new VorbisTrackInfo(trackName[i]); + if (!trackInfo->error()) + return trackInfo; + delete trackInfo; + } + } + return NULL; +} + + + } // End of namespace Audio #endif // #ifdef USE_VORBIS This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.

[Scummvm-cvs-logs] SF.net SVN: scummvm: [25784] scummvm/trunk/sound From: <fingolfin@us...> - 2007-02-22 10:53 Revision: 25784 http://scummvm.svn.sourceforge.net/scummvm/?rev=25784&view=rev Author: fingolfin Date: 2007-02-22 02:51:47 -0800 (Thu, 22 Feb 2007) Log Message: ----------- Rewrote Ogg Vorbis code to be more flexible when it comes to seeking; also now playback from arbitrary SeekableReadStream data sources is possible Modified Paths: -------------- scummvm/trunk/sound/mp3.cpp scummvm/trunk/sound/vorbis.cpp Modified: scummvm/trunk/sound/mp3.cpp =================================================================== --- scummvm/trunk/sound/mp3.cpp 2007-02-22 10:32:06 UTC (rev 25783) +++ scummvm/trunk/sound/mp3.cpp 2007-02-22 10:51:47 UTC (rev 25784) @@ -299,10 +299,10 @@ if (!size) size = file->size() - file->pos(); - // Read 'size' bytes of data (or until EOF is reached) into a MemoryReadStream + // Read 'size' bytes of data into a MemoryReadStream Common::MemoryReadStream *stream = file->readStream(size); - // .. and create a MP3InputStream from all this + // .. and create an MP3InputStream from all this return new MP3InputStream(stream, true); } Modified: scummvm/trunk/sound/vorbis.cpp =================================================================== --- scummvm/trunk/sound/vorbis.cpp 2007-02-22 10:32:06 UTC (rev 25783) +++ scummvm/trunk/sound/vorbis.cpp 2007-02-22 10:51:47 UTC (rev 25784) @@ -46,166 +46,141 @@ namespace Audio { -// These are wrapper functions to allow using a File object to +// These are wrapper functions to allow using a SeekableReadStream object to // provide data to the OggVorbis_File object. -struct file_info { - File *file; - int start, curr_pos; - size_t len; -}; +static size_t read_stream_wrap(void *ptr, size_t size, size_t nmemb, void *datasource) { + Common::SeekableReadStream *stream = (Common::SeekableReadStream *)datasource; -static size_t read_wrap(void *ptr, size_t size, size_t nmemb, void *datasource) { - file_info *f = (file_info *) datasource; - int result; - -#ifdef __SYMBIAN32__ - // For symbian we must check that an alternative file pointer is created, see if its open - // If not re-open file and seek to the last read position - if (f->file && !f->file->isOpen()) { - f->file->open(f->file->name()); - f->file->seek(f->curr_pos); - } -#endif - - nmemb *= size; - if (f->curr_pos > (int) f->len) - nmemb = 0; - else if (nmemb > f->len - f->curr_pos) - nmemb = f->len - f->curr_pos; - // There is no guarantee that the Vorbis stream is alone in accessing - // the file, so make sure the current position is what we think it is. - f->file->seek(f->start + f->curr_pos); - result = f->file->read(ptr, nmemb); -#ifdef __SYMBIAN32__ - // For symbian we now store the last read position and then close the file - if (f->file) { - f->file->close(); - } -#endif - if (result == -1) { - f->curr_pos = f->file->pos() - f->start; - return (size_t) -1; - } else { - f->curr_pos += result; - return result / size; - } + uint32 result = stream->read(ptr, size * nmemb); + + return result / size; } -static int seek_wrap(void *datasource, ogg_int64_t offset, int whence) { - file_info *f = (file_info *) datasource; - - if (whence == SEEK_SET) - offset += f->start; - else if (whence == SEEK_END) { - offset += f->start + f->len; - whence = SEEK_SET; - } - -#ifdef __SYMBIAN32__ - // For symbian we must check that an alternative file pointer is created, see if its open - // If not re-open file and seek to the last read position - if (f->file && !f->file->isOpen()) { - f->file->open(f->file->name()); - f->file->seek(f->curr_pos); - } -#endif - - f->file->seek(offset, whence); - f->curr_pos = f->file->pos() - f->start; - -#ifdef __SYMBIAN32__ - // For symbian we now store the last read position and then close the file - if (f->file) { - f->file->close(); - } -#endif - - return f->curr_pos; +static int seek_stream_wrap(void *datasource, ogg_int64_t offset, int whence) { + Common::SeekableReadStream *stream = (Common::SeekableReadStream *)datasource; + stream->seek(offset, whence); + return stream->pos(); } -static int close_wrap(void *datasource) { - file_info *f = (file_info *) datasource; - - delete f->file; - delete f; - +static int close_stream_wrap(void *datasource) { + // Do nothing -- we leave it up to the VorbisInputStream to free memory as appropriate. return 0; } -static long tell_wrap(void *datasource) { - file_info *f = (file_info *) datasource; - - return f->curr_pos; +static long tell_stream_wrap(void *datasource) { + Common::SeekableReadStream *stream = (Common::SeekableReadStream *)datasource; + return stream->pos(); } -static ov_callbacks g_File_wrap = { - read_wrap, seek_wrap, close_wrap, tell_wrap +static ov_callbacks g_stream_wrap = { + read_stream_wrap, seek_stream_wrap, close_stream_wrap, tell_stream_wrap }; + #pragma mark - #pragma mark --- Ogg Vorbis stream --- #pragma mark - class VorbisInputStream : public AudioStream { - OggVorbis_File *_ov_file; - int _end_pos; - int _numChannels; +protected: + OggVorbis_File _ovFile; + + Common::SeekableReadStream *_inStream; + bool _disposeAfterUse; + int16 _buffer[4096]; const int16 *_bufferEnd; const int16 *_pos; - bool _deleteFileAfterUse; + + bool _isStereo; + int _rate; + +#ifdef USE_TREMOR + ogg_int64_t _startTime; + ogg_int64_t _endTime; +#else + double _startTime; + double _endTime; +#endif void refill(); inline bool eosIntern() const; public: - VorbisInputStream(OggVorbis_File *file, int duration, bool deleteFileAfterUse); + // startTime / duration are in milliseconds + VorbisInputStream(Common::SeekableReadStream *inStream, bool dispose, uint startTime = 0, uint endTime = 0); ~VorbisInputStream(); int readBuffer(int16 *buffer, const int numSamples); bool endOfData() const { return eosIntern(); } - bool isStereo() const { return _numChannels >= 2; } + bool isStereo() const { return _isStereo; } - int getRate() const { return ov_info(_ov_file, -1)->rate; } + int getRate() const { return _rate; } }; -VorbisInputStream::VorbisInputStream(OggVorbis_File *file, int duration, bool deleteFileAfterUse) - : _ov_file(file), - _bufferEnd(_buffer + ARRAYSIZE(_buffer)), - _deleteFileAfterUse(deleteFileAfterUse) { -//debug(5, "" __FILE__ ":%i", __LINE__); +VorbisInputStream::VorbisInputStream(Common::SeekableReadStream *inStream, bool dispose, uint startTime, uint endTime) : + _inStream(inStream), + _disposeAfterUse(dispose), + _bufferEnd(_buffer + ARRAYSIZE(_buffer)) { - // Check the header, determine if this is a stereo stream - _numChannels = ov_info(_ov_file, -1)->channels; + bool err = (ov_open_callbacks(inStream, &_ovFile, NULL, 0, g_stream_wrap) < 0); + // FIXME: proper error handling! + assert(!err); - // Determine the end position - if (duration) - _end_pos = ov_pcm_tell(_ov_file) + duration; - else - _end_pos = ov_pcm_total(_ov_file, -1); +#ifdef USE_TREMOR + /* TODO: Symbian may have to use scumm_fixdfdi here? To quote: + "SumthinWicked says: fixing "relocation truncated to fit: ARM_26 __fixdfdi" during linking on GCC, see portdefs.h" + */ + // In Tremor, the ov_time_seek() and ov_time_seek_page() calls take seeking + // positions in milliseconds as 64 bit integers, rather than in seconds as + // doubles as in Vorbisfile. + ogg_int64_t totalTime; + _startTime = startTime; + _endTime = endTime; +#else + double totalTime; + _startTime = startTime / 1000.0; + _endTime = endTime / 1000.0; +#endif + + // If endTime was 0, or is past the end of the file, set it to the maximal time possible + totalTime = ov_time_total(&_ovFile, -1); + if (_endTime == 0 || _endTime > totalTime) + _endTime = totalTime; + + // If the specified time range is empty, abort early. + if (_startTime >= _endTime) { + _pos = _bufferEnd; + return; + } + + // Seek to the start position + ov_time_seek(&_ovFile, _startTime); + // Read in initial data refill(); + + // Setup some header information + _isStereo = ov_info(&_ovFile, -1)->channels >= 2; + _rate = ov_info(&_ovFile, -1)->rate; } VorbisInputStream::~VorbisInputStream() { -//debug(5, "" __FILE__ ":%i", __LINE__); - ov_clear(_ov_file); - if (_deleteFileAfterUse) - delete _ov_file; + ov_clear(&_ovFile); + if (_disposeAfterUse) + delete _inStream; } inline bool VorbisInputStream::eosIntern() const { -//debug(5, "" __FILE__ ":%i", __LINE__); return _pos >= _bufferEnd; } int VorbisInputStream::readBuffer(int16 *buffer, const int numSamples) { -//debug(5, "" __FILE__ ":%i", __LINE__); int samples = 0; while (samples < numSamples && !eosIntern()) { const int len = MIN(numSamples - samples, (int)(_bufferEnd - _pos)); @@ -221,23 +196,33 @@ } void VorbisInputStream::refill() { -//debug(5, "" __FILE__ ":%i", __LINE__); // 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 USE_TREMOR // Tremor ov_read() always returns data as signed 16 bit interleaved PCM in host byte order. As such, it does not take arguments to request specific signedness, byte order or bit depth as in Vorbisfile. + while (len_left > 0 && ov_time_tell(&_ovFile) < _endTime) { + long result; +#ifdef USE_TREMOR + // Tremor ov_read() always returns data as signed 16 bit interleaved PCM + // in host byte order. As such, it does not take arguments to request + // specific signedness, byte order or bit depth as in Vorbisfile. + result = ov_read(&_ovFile, read_pos, len_left, + NULL); +#else #ifdef SCUMM_BIG_ENDIAN + result = ov_read(&_ovFile, read_pos, len_left, 1, + 2, // 16 bit + 1, // signed + NULL); #else + result = ov_read(&_ovFile, read_pos, len_left, 0, -#endif 2, // 16 bit 1, // signed -#endif NULL); +#endif +#endif if (result == OV_HOLE) { // Possibly recoverable, just warn about it warning("Corrupted data in Vorbis file"); @@ -259,34 +244,20 @@ } AudioStream *makeVorbisStream(File *file, uint32 size) { -//debug(5, "" __FILE__ ":%i", __LINE__); - OggVorbis_File *ov_file = new OggVorbis_File; - file_info *f = new file_info; + assert(file); -#if defined(__SYMBIAN32__) - // Symbian can't share filehandles between different threads. - // So create a new file and seek that to the other filehandles position - f->file = new File; - f->file->open(file->name()); - f->file->seek(file->pos()); -#else - f->file = file; -#endif - f->start = file->pos(); - f->len = size; - f->curr_pos = 0; + // FIXME: For now, just read the whole data into memory, and be done + // with it. Of course this is in general *not* a nice thing to do... - if (ov_open_callbacks((void *) f, ov_file, NULL, 0, g_File_wrap) < 0) { - warning("Invalid file format"); - delete ov_file; - delete f; - return 0; - } else { -#ifndef __SYMBIAN32__ - file->incRef(); -#endif - return new VorbisInputStream(ov_file, 0, true); - } + // If no size was specified, read the whole remainder of the file + if (!size) + size = file->size() - file->pos(); + + // Read 'size' bytes of data into a MemoryReadStream + Common::MemoryReadStream *stream = file->readStream(size); + + // .. and create a VorbisInputStream from all this + return new VorbisInputStream(stream, true); } @@ -301,7 +272,6 @@ public: VorbisTrackInfo(const char *filename); - bool openTrack(OggVorbis_File *ovFile); bool error() { return _errorFlag; } void play(Audio::Mixer *mixer, Audio::SoundHandle *handle, int startFrame, int duration); }; @@ -310,60 +280,46 @@ VorbisTrackInfo::VorbisTrackInfo(const char *filename) : _filename(filename), _errorFlag(false) { - - OggVorbis_File ov_file; - _errorFlag = openTrack(&ov_file); - if (!_errorFlag) - ov_clear(&ov_file); -} -bool VorbisTrackInfo::openTrack(OggVorbis_File *ovFile) { - bool err; - - file_info *f = new file_info; - assert(f); - f->file = new File; - assert(f->file); - - err = !f->file->open(_filename); - - if (!err) { - f->start = 0; - f->len = f->file->size(); - f->curr_pos = 0; - - err = (ov_open_callbacks((void *) f, ovFile, NULL, 0, g_File_wrap) < 0); + // Try to open the file + Common::File file; + if (!file.open(_filename)) { + _errorFlag = true; + return; } + + // Next, try to create a VorbisInputStream from it + VorbisInputStream *tempStream = new VorbisInputStream(&file, false); - if (err) { - delete f->file; - delete f; - } - - return err; + // If an error occured... + // TODO: add an error or init method to VorbisInputStream + _errorFlag = tempStream->endOfData(); + + // Clean up again + delete tempStream; } void VorbisTrackInfo::play(Audio::Mixer *mixer, Audio::SoundHandle *handle, int startFrame, int duration) { - OggVorbis_File *ovFile = new OggVorbis_File; - bool err = openTrack(ovFile); - assert(!err); + assert(!_errorFlag); + uint start, end; -#ifdef USE_TREMOR - // In Tremor, the ov_time_seek() and ov_time_seek_page() calls take seeking - // positions in milliseconds as 64 bit integers, rather than in seconds as - // doubles as in Vorbisfile. -#if defined(__SYMBIAN32__) && defined(__GCC32__) - // SumthinWicked says: fixing "relocation truncated to fit: ARM_26 __fixdfdi" during linking on GCC, see portdefs.h - ov_time_seek(ovFile, (ogg_int64_t)scumm_fixdfdi(startFrame / 75.0 * 1000)); -#else - ov_time_seek(ovFile, (ogg_int64_t)(startFrame / 75.0 * 1000)); -#endif -#else - ov_time_seek(ovFile, startFrame / 75.0); -#endif + // Open the file + Common::File *file = new Common::File(); + if (!file || !file->open(_filename)) { + warning("VorbisTrackInfo::play: failed to open '%s'", _filename.c_str()); + delete file; + return; + } + + // Convert startFrame & duration from frames (1/75 s) to milliseconds (1/1000s) + start = startFrame * 1000 / 75; + end = duration ? ((startFrame + duration) * 1000 / 75) : 0; - AudioStream *input = new VorbisInputStream(ovFile, duration * ov_info(ovFile, -1)->rate / 75, true); + // ... create an AudioStream ... + VorbisInputStream *input = new VorbisInputStream(file, true, start, end); + + // ... and play it mixer->playInputStream(Audio::Mixer::kMusicSoundType, handle, input); } This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.

[Scummvm-cvs-logs] SF.net SVN: scummvm: [25789] scummvm/trunk/sound From: <fingolfin@us...> - 2007-02-22 14:30 Revision: 25789 http://scummvm.svn.sourceforge.net/scummvm/?rev=25789&view=rev Author: fingolfin Date: 2007-02-22 06:30:12 -0800 (Thu, 22 Feb 2007) Log Message: ----------- Added looping support and an enhanced factory function to the Vorbis code Modified Paths: -------------- scummvm/trunk/sound/mp3.cpp scummvm/trunk/sound/vorbis.cpp scummvm/trunk/sound/vorbis.h Modified: scummvm/trunk/sound/mp3.cpp =================================================================== --- scummvm/trunk/sound/mp3.cpp 2007-02-22 14:07:12 UTC (rev 25788) +++ scummvm/trunk/sound/mp3.cpp 2007-02-22 14:30:12 UTC (rev 25789) @@ -208,10 +208,11 @@ break; } - if (_eos) { - // If looping is enabled, rewind to the start - if (_numLoops == 0 || --_numLoops > 0) - rewind(); + if (_eos && _numLoops != 1) { + // If looping is on and there are loops left, rewind to the start + if (_numLoops != 0) + _numLoops--; + rewind(); } } while (_stream.error == MAD_ERROR_BUFLEN); Modified: scummvm/trunk/sound/vorbis.cpp =================================================================== --- scummvm/trunk/sound/vorbis.cpp 2007-02-22 14:07:12 UTC (rev 25788) +++ scummvm/trunk/sound/vorbis.cpp 2007-02-22 14:30:12 UTC (rev 25789) @@ -86,17 +86,12 @@ class VorbisInputStream : public AudioStream { protected: - OggVorbis_File _ovFile; - Common::SeekableReadStream *_inStream; bool _disposeAfterUse; - int16 _buffer[4096]; - const int16 *_bufferEnd; - const int16 *_pos; - bool _isStereo; int _rate; + uint _numLoops; #ifdef USE_TREMOR ogg_int64_t _startTime; @@ -106,25 +101,31 @@ double _endTime; #endif - void refill(); - inline bool eosIntern() const; + OggVorbis_File _ovFile; + + int16 _buffer[4096]; + const int16 *_bufferEnd; + const int16 *_pos; + public: // startTime / duration are in milliseconds - VorbisInputStream(Common::SeekableReadStream *inStream, bool dispose, uint startTime = 0, uint endTime = 0); + VorbisInputStream(Common::SeekableReadStream *inStream, bool dispose, uint startTime = 0, uint endTime = 0, uint numLoops = 1); ~VorbisInputStream(); int readBuffer(int16 *buffer, const int numSamples); - bool endOfData() const { return eosIntern(); } + bool endOfData() const { return _pos >= _bufferEnd; } bool isStereo() const { return _isStereo; } - int getRate() const { return _rate; } +protected: + void refill(); }; -VorbisInputStream::VorbisInputStream(Common::SeekableReadStream *inStream, bool dispose, uint startTime, uint endTime) : +VorbisInputStream::VorbisInputStream(Common::SeekableReadStream *inStream, bool dispose, uint startTime, uint endTime, uint numLoops) : _inStream(inStream), _disposeAfterUse(dispose), + _numLoops(numLoops), _bufferEnd(_buffer + ARRAYSIZE(_buffer)) { bool err = (ov_open_callbacks(inStream, &_ovFile, NULL, 0, g_stream_wrap) < 0); @@ -176,13 +177,9 @@ delete _inStream; } -inline bool VorbisInputStream::eosIntern() const { - return _pos >= _bufferEnd; -} - int VorbisInputStream::readBuffer(int16 *buffer, const int numSamples) { int samples = 0; - while (samples < numSamples && !eosIntern()) { + while (samples < numSamples && _pos < _bufferEnd) { const int len = MIN(numSamples - samples, (int)(_bufferEnd - _pos)); memcpy(buffer, _pos, len * 2); buffer += len; @@ -190,6 +187,17 @@ samples += len; if (_pos >= _bufferEnd) { refill(); + // If we are still out of data, and also past the end of specified + // time range, check whether looping is enabled... + if (_pos >= _bufferEnd && ov_time_tell(&_ovFile) >= _endTime) { + if (_numLoops != 1) { + // If looping is on and there are loops left, rewind to the start + if (_numLoops != 0) + _numLoops--; + ov_time_seek(&_ovFile, _startTime); + refill(); + } + } } } return samples; @@ -200,7 +208,16 @@ uint len_left = sizeof(_buffer); char *read_pos = (char *)_buffer; - while (len_left > 0 && ov_time_tell(&_ovFile) < _endTime) { + while (len_left > 0) { + if (ov_time_tell(&_ovFile) >= _endTime) { + // If looping is on and there are loops left, rewind to the start + if (_numLoops == 1) + break; // Last loop, abort + if (_numLoops != 0) + _numLoops--; + ov_time_seek(&_ovFile, _startTime); + } + long result; #ifdef USE_TREMOR // Tremor ov_read() always returns data as signed 16 bit interleaved PCM @@ -260,7 +277,20 @@ return new VorbisInputStream(stream, true); } +AudioStream *makeVorbisStream( + Common::SeekableReadStream *stream, + bool disposeAfterUse, + uint32 startTime, + uint32 duration, + uint numLoops) { + uint32 endTime = duration ? (startTime + duration) : 0; + + return new VorbisInputStream(stream, disposeAfterUse, startTime, endTime, numLoops); +} + + + #pragma mark - #pragma mark --- Ogg Vorbis Audio CD emulation --- #pragma mark - Modified: scummvm/trunk/sound/vorbis.h =================================================================== --- scummvm/trunk/sound/vorbis.h 2007-02-22 14:07:12 UTC (rev 25788) +++ scummvm/trunk/sound/vorbis.h 2007-02-22 14:30:12 UTC (rev 25789) @@ -30,6 +30,7 @@ namespace Common { class File; + class SeekableReadStream; } namespace Audio { @@ -40,13 +41,34 @@ DigitalTrackInfo *getVorbisTrack(int track); /** - * Create a new AudioStream from the Vorbis data in the given + * Create a new AudioStream from the Ogg Vorbis data in the given * file. If you only want to play part of that file, then seek * to the start position in file before passing it to this * factory function, and specify the appropriate size. */ AudioStream *makeVorbisStream(Common::File *file, uint32 size); + +/** + * Create a new AudioStream from the Ogg Vorbis data in the given stream. + * Allows for looping (which is why we require a SeekableReadStream), + * and specifying only a portion of the data to be played, based + * on time offsets. + * + * @param stream the SeekableReadStream from which to read the MP3 data + * @param disposeAfterUse whether to delete the stream after use + * @param startTime the (optional) time offset in milliseconds from which to start playback + * @param duration the (optional) time in milliseconds specifying how long to play + * @param numLoops how often the data shall be looped (0 = infinite) + * @return a new AudioStream, or NULL, if an error occured + */ +AudioStream *makeVorbisStream( + Common::SeekableReadStream *stream, + bool disposeAfterUse, + uint32 startTime = 0, + uint32 duration = 0, + uint numLoops = 1); + } // End of namespace Audio #endif // #ifdef USE_VORBIS This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.

[Scummvm-cvs-logs] SF.net SVN: scummvm: [25792] scummvm/trunk/sound From: <fingolfin@us...> - 2007-02-22 18:21 Revision: 25792 http://scummvm.svn.sourceforge.net/scummvm/?rev=25792&view=rev Author: fingolfin Date: 2007-02-22 10:21:01 -0800 (Thu, 22 Feb 2007) Log Message: ----------- Updated FLAC code to match the changes made to the MP3 & Vorbis decoders (e.g. use a SeekableReadStream for input, allow specifying parts of a stream via time-valued parameters, etc.) -- note: looping is not yet implemented, even though it is already present in the API Modified Paths: -------------- scummvm/trunk/sound/flac.cpp scummvm/trunk/sound/flac.h scummvm/trunk/sound/vorbis.cpp Modified: scummvm/trunk/sound/flac.cpp =================================================================== --- scummvm/trunk/sound/flac.cpp 2007-02-22 18:03:05 UTC (rev 25791) +++ scummvm/trunk/sound/flac.cpp 2007-02-22 18:21:01 UTC (rev 25792) @@ -82,61 +82,14 @@ class FlacInputStream : public AudioStream { - -public: - FlacInputStream(File *sourceFile, const uint32 fileStart, const uint32 fileStop); - FlacInputStream(File *sourceFile, const uint32 fileStart = 0); - virtual ~FlacInputStream(); - - int readBuffer(int16 *buffer, const int numSamples); - - bool isStereo() const { return _streaminfo.channels >= 2; } - int getRate() const { return _streaminfo.sample_rate; } - bool endOfStream() const { return _streaminfo.channels == 0 || (_lastSampleWritten && getBufferedSamples() == 0); } - /** the mixer aint supporting it right now.. */ - //bool endOfData() const { return getBufferedSamples() == 0; } - bool endOfData() const { return endOfStream(); } - - uint getChannels() const { return MIN(_streaminfo.channels, MAX_OUTPUT_CHANNELS); } - uint getBufferedSamples() const { return _preBuffer.bufFill; }; - - const FLAC__StreamMetadata_StreamInfo& getStreamInfo() const {return _streaminfo;} - - inline FLAC__SeekableStreamDecoderState getState() const; - inline FLAC__StreamDecoderState getStreamDecoderState() const; - - - bool isStreamDecoderReady() const { return getStreamDecoderState() == FLAC__STREAM_DECODER_SEARCH_FOR_FRAME_SYNC ; } - bool init(); - bool finish(); - bool flush(); - inline bool processSingleBlock(); - inline bool processUntilEndOfMetadata(); - bool seekAbsolute(FLAC__uint64 sample); - inline void setLastSample(FLAC__uint64 absoluteSample); - protected: - inline ::FLAC__SeekableStreamDecoderReadStatus callbackRead(FLAC__byte buffer[], FLAC_size_t *bytes); - inline ::FLAC__SeekableStreamDecoderSeekStatus callbackSeek(FLAC__uint64 absoluteByteOffset); - inline ::FLAC__SeekableStreamDecoderTellStatus callbackTell(FLAC__uint64 *absoluteByteOffset); - inline ::FLAC__SeekableStreamDecoderLengthStatus callbackLength(FLAC__uint64 *streamLength); - inline bool callbackEOF(); - inline ::FLAC__StreamDecoderWriteStatus callbackWrite(const ::FLAC__Frame *frame, const FLAC__int32 * const buffer[]); - inline void callbackMetadata(const ::FLAC__StreamMetadata *metadata); - inline void callbackError(::FLAC__StreamDecoderErrorStatus status); + Common::SeekableReadStream *_inStream; + bool _disposeAfterUse; + uint _numLoops; + ::FLAC__SeekableStreamDecoder *_decoder; -private: - static ::FLAC__SeekableStreamDecoderReadStatus callWrapRead(const ::FLAC__SeekableStreamDecoder *decoder, FLAC__byte buffer[], FLAC_size_t *bytes, void *clientData); - static ::FLAC__SeekableStreamDecoderSeekStatus callWrapSeek(const ::FLAC__SeekableStreamDecoder *decoder, FLAC__uint64 absoluteByteOffset, void *clientData); - static ::FLAC__SeekableStreamDecoderTellStatus callWrapTell(const ::FLAC__SeekableStreamDecoder *decoder, FLAC__uint64 *absoluteByteOffset, void *clientData); - static ::FLAC__SeekableStreamDecoderLengthStatus callWrapLength(const ::FLAC__SeekableStreamDecoder *decoder, FLAC__uint64 *streamLength, void *clientData); - static FLAC__bool callWrapEOF(const ::FLAC__SeekableStreamDecoder *decoder, void *clientData); - static ::FLAC__StreamDecoderWriteStatus callWrapWrite(const ::FLAC__SeekableStreamDecoder *decoder, const ::FLAC__Frame *frame, const FLAC__int32 * const buffer[], void *clientData); - static void callWrapMetadata(const ::FLAC__SeekableStreamDecoder *decoder, const ::FLAC__StreamMetadata *metadata, void *clientData); - static void callWrapError(const ::FLAC__SeekableStreamDecoder *decoder, ::FLAC__StreamDecoderErrorStatus status, void *clientData); - // Private and undefined so you can't use them: FlacInputStream(const FlacInputStream &); void operator=(const FlacInputStream &); @@ -150,19 +103,9 @@ /** Header of the Stream */ FLAC__StreamMetadata_StreamInfo _streaminfo; - struct { - /** Handle to the File */ - File *fileHandle; - /** Index of next Byte to read */ - uint32 filePos; - /** start of stream - not necessary start of file */ - uint32 fileStartPos; - /** last index of Stream + 1(!) - not necessary end of file */ - uint32 fileEndPos; - } _fileInfo; - /** index of the first Sample to be played */ FLAC__uint64 _firstSample; + /** index + 1(!) of the last Sample to be played - 0 is end of Stream*/ FLAC__uint64 _lastSample; @@ -182,9 +125,52 @@ bufType *_outBuffer; uint _requestedSamples; - void setBestConvertBufferMethod(); typedef void (*PFCONVERTBUFFERS)(bufType*,const FLAC__int32*[], uint, const uint, const uint8); PFCONVERTBUFFERS _methodConvertBuffers; + + +public: + FlacInputStream(Common::SeekableReadStream *inStream, bool dispose, uint startTime = 0, uint endTime = 0, uint numLoops = 1); + virtual ~FlacInputStream(); + + int readBuffer(int16 *buffer, const int numSamples); + + bool isStereo() const { return _streaminfo.channels >= 2; } + int getRate() const { return _streaminfo.sample_rate; } + bool endOfData() const { return _streaminfo.channels == 0 || (_lastSampleWritten && _preBuffer.bufFill == 0); } + + bool isStreamDecoderReady() const { return getStreamDecoderState() == FLAC__STREAM_DECODER_SEARCH_FOR_FRAME_SYNC ; } + +protected: + uint getChannels() const { return MIN(_streaminfo.channels, MAX_OUTPUT_CHANNELS); } + + inline FLAC__SeekableStreamDecoderState getState() const; + inline FLAC__StreamDecoderState getStreamDecoderState() const; + + inline bool processSingleBlock(); + inline bool processUntilEndOfMetadata(); + bool seekAbsolute(FLAC__uint64 sample); + + inline ::FLAC__SeekableStreamDecoderReadStatus callbackRead(FLAC__byte buffer[], FLAC_size_t *bytes); + inline ::FLAC__SeekableStreamDecoderSeekStatus callbackSeek(FLAC__uint64 absoluteByteOffset); + inline ::FLAC__SeekableStreamDecoderTellStatus callbackTell(FLAC__uint64 *absoluteByteOffset); + inline ::FLAC__SeekableStreamDecoderLengthStatus callbackLength(FLAC__uint64 *streamLength); + inline bool callbackEOF(); + inline ::FLAC__StreamDecoderWriteStatus callbackWrite(const ::FLAC__Frame *frame, const FLAC__int32 * const buffer[]); + inline void callbackMetadata(const ::FLAC__StreamMetadata *metadata); + inline void callbackError(::FLAC__StreamDecoderErrorStatus status); + +private: + static ::FLAC__SeekableStreamDecoderReadStatus callWrapRead(const ::FLAC__SeekableStreamDecoder *decoder, FLAC__byte buffer[], FLAC_size_t *bytes, void *clientData); + static ::FLAC__SeekableStreamDecoderSeekStatus callWrapSeek(const ::FLAC__SeekableStreamDecoder *decoder, FLAC__uint64 absoluteByteOffset, void *clientData); + static ::FLAC__SeekableStreamDecoderTellStatus callWrapTell(const ::FLAC__SeekableStreamDecoder *decoder, FLAC__uint64 *absoluteByteOffset, void *clientData); + static ::FLAC__SeekableStreamDecoderLengthStatus callWrapLength(const ::FLAC__SeekableStreamDecoder *decoder, FLAC__uint64 *streamLength, void *clientData); + static FLAC__bool callWrapEOF(const ::FLAC__SeekableStreamDecoder *decoder, void *clientData); + static ::FLAC__StreamDecoderWriteStatus callWrapWrite(const ::FLAC__SeekableStreamDecoder *decoder, const ::FLAC__Frame *frame, const FLAC__int32 * const buffer[], void *clientData); + static void callWrapMetadata(const ::FLAC__SeekableStreamDecoder *decoder, const ::FLAC__StreamMetadata *metadata, void *clientData); + static void callWrapError(const ::FLAC__SeekableStreamDecoder *decoder, ::FLAC__StreamDecoderErrorStatus status, void *clientData); + + void setBestConvertBufferMethod(); static void convertBuffersGeneric(bufType* bufDestination, const FLAC__int32 *inChannels[], uint numSamples, const uint numChannels, const uint8 numBits); static void convertBuffersStereoNS(bufType* bufDestination, const FLAC__int32 *inChannels[], uint numSamples, const uint numChannels, const uint8 numBits); static void convertBuffersStereo8Bit(bufType* bufDestination, const FLAC__int32 *inChannels[], uint numSamples, const uint numChannels, const uint8 numBits); @@ -192,102 +178,33 @@ static void convertBuffersMono8Bit(bufType* bufDestination, const FLAC__int32 *inChannels[], uint numSamples, const uint numChannels, const uint8 numBits); }; -FlacInputStream::FlacInputStream(File *sourceFile, const uint32 fileStart) +FlacInputStream::FlacInputStream(Common::SeekableReadStream *inStream, bool dispose, uint startTime, uint endTime, uint numLoops) #ifdef LEGACY_FLAC : _decoder(::FLAC__seekable_stream_decoder_new()), #else : _decoder(::FLAC__stream_decoder_new()), #endif - _firstSample(0), _lastSample(0), - _outBuffer(NULL), _requestedSamples(0), _lastSampleWritten(true), - _methodConvertBuffers(&FlacInputStream::convertBuffersGeneric) + _inStream(inStream), + _disposeAfterUse(dispose), + _numLoops(numLoops), + _firstSample(0), _lastSample(0), + _outBuffer(NULL), _requestedSamples(0), _lastSampleWritten(true), + _methodConvertBuffers(&FlacInputStream::convertBuffersGeneric) { - assert(sourceFile != NULL && sourceFile->isOpen()); - + assert(_inStream); memset(&_streaminfo, 0, sizeof(_streaminfo)); - _preBuffer.bufData = NULL; - _preBuffer.bufFill = 0; - _preBuffer.bufSize = 0; +// TODO: Implement looping support - _fileInfo.fileHandle = sourceFile; - _fileInfo.fileStartPos = fileStart; - _fileInfo.filePos = fileStart; - _fileInfo.fileEndPos = sourceFile->size(); - - _fileInfo.fileHandle->incRef(); -} - -FlacInputStream::FlacInputStream(File *sourceFile, const uint32 fileStart, const uint32 fileStop) -#ifdef LEGACY_FLAC - : _decoder(::FLAC__seekable_stream_decoder_new()), -#else - : _decoder(::FLAC__stream_decoder_new()), -#endif - _firstSample(0), _lastSample(0), - _outBuffer(NULL), _requestedSamples(0), _lastSampleWritten(true), - _methodConvertBuffers(&FlacInputStream::convertBuffersGeneric) -{ - assert(sourceFile != NULL && sourceFile->isOpen()); - assert(fileStop <= 0 || (fileStart < fileStop && fileStop <= sourceFile->size())); - - memset(&_streaminfo, 0, sizeof(_streaminfo)); - _preBuffer.bufData = NULL; _preBuffer.bufFill = 0; _preBuffer.bufSize = 0; - _fileInfo.fileHandle = sourceFile; - _fileInfo.fileStartPos = fileStart; - _fileInfo.filePos = fileStart; - _fileInfo.fileEndPos = fileStop; - - _fileInfo.fileHandle->incRef(); -} - -FlacInputStream::~FlacInputStream() { - if (_decoder != NULL) { -#ifdef LEGACY_FLAC - (void) ::FLAC__seekable_stream_decoder_finish(_decoder); - ::FLAC__seekable_stream_decoder_delete(_decoder); -#else - (void) ::FLAC__stream_decoder_finish(_decoder); - ::FLAC__stream_decoder_delete(_decoder); -#endif - } - if (_preBuffer.bufData != NULL) - delete[] _preBuffer.bufData; - - _fileInfo.fileHandle->decRef(); -} - -inline FLAC__SeekableStreamDecoderState FlacInputStream::getState() const { - assert(isValid()); -#ifdef LEGACY_FLAC - return ::FLAC__seekable_stream_decoder_get_state(_decoder); -#else - return ::FLAC__stream_decoder_get_state(_decoder); -#endif -} - -inline FLAC__StreamDecoderState FlacInputStream::getStreamDecoderState() const { - assert(isValid()); -#ifdef LEGACY_FLAC - return ::FLAC__seekable_stream_decoder_get_stream_decoder_state(_decoder); -#else - return ::FLAC__stream_decoder_get_state(_decoder); -#endif -} - -bool FlacInputStream::init() { - assert(isValid()); - - memset(&_streaminfo, 0, sizeof (_streaminfo)); deleteBuffer(); - _fileInfo.filePos = _fileInfo.fileStartPos; _lastSampleWritten = false; _methodConvertBuffers = &FlacInputStream::convertBuffersGeneric; + bool success; #ifdef LEGACY_FLAC ::FLAC__seekable_stream_decoder_set_read_callback(_decoder, &FlacInputStream::callWrapRead); ::FLAC__seekable_stream_decoder_set_seek_callback(_decoder, &FlacInputStream::callWrapSeek); @@ -299,16 +216,9 @@ ::FLAC__seekable_stream_decoder_set_error_callback(_decoder, &FlacInputStream::callWrapError); ::FLAC__seekable_stream_decoder_set_client_data(_decoder, (void*)this); - if (::FLAC__seekable_stream_decoder_init(_decoder) == FLAC__SEEKABLE_STREAM_DECODER_OK) { - if (processUntilEndOfMetadata() && _streaminfo.channels > 0) { - if (_firstSample == 0 || 0 != ::FLAC__seekable_stream_decoder_seek_absolute(_decoder, _firstSample)) { - // FLAC__StreamDecoderState state = getStreamDecoderState(); - return true; // no error occured - } - } - } + success = (::FLAC__seekable_stream_decoder_init(_decoder) == FLAC__SEEKABLE_STREAM_DECODER_OK); #else - if (::FLAC__stream_decoder_init_stream( + success = (::FLAC__stream_decoder_init_stream( _decoder, &FlacInputStream::callWrapRead, &FlacInputStream::callWrapSeek, @@ -319,37 +229,52 @@ &FlacInputStream::callWrapMetadata, &FlacInputStream::callWrapError, (void*)this - ) == FLAC__STREAM_DECODER_INIT_STATUS_OK) { + ) == FLAC__STREAM_DECODER_INIT_STATUS_OK); +#endif + if (success) { if (processUntilEndOfMetadata() && _streaminfo.channels > 0) { - if (_firstSample == 0 || 0 != ::FLAC__stream_decoder_seek_absolute(_decoder, _firstSample)) { - // FLAC__StreamDecoderState state = getStreamDecoderState(); - return true; // no error occured + _firstSample = (FLAC__uint64)(startTime * (_streaminfo.sample_rate / 1000.0)); + _lastSample = (FLAC__uint64)(endTime * (_streaminfo.sample_rate / 1000.0)); + if (_firstSample == 0 || seekAbsolute(_firstSample)) { + return; // no error occured } } } + + warning("FlacInputStream: could not create audio stream"); +} + +FlacInputStream::~FlacInputStream() { + if (_decoder != NULL) { +#ifdef LEGACY_FLAC + (void) ::FLAC__seekable_stream_decoder_finish(_decoder); + ::FLAC__seekable_stream_decoder_delete(_decoder); +#else + (void) ::FLAC__stream_decoder_finish(_decoder); + ::FLAC__stream_decoder_delete(_decoder); #endif + } + delete[] _preBuffer.bufData; - warning("FlacInputStream: could not create an Audiostream from File %s", _fileInfo.fileHandle->name()); - return false; + if (_disposeAfterUse) + delete _inStream; } -bool FlacInputStream::finish() { +inline FLAC__SeekableStreamDecoderState FlacInputStream::getState() const { assert(isValid()); - deleteBuffer(); #ifdef LEGACY_FLAC - return 0 != ::FLAC__seekable_stream_decoder_finish(_decoder); + return ::FLAC__seekable_stream_decoder_get_state(_decoder); #else - return 0 != ::FLAC__stream_decoder_finish(_decoder); + return ::FLAC__stream_decoder_get_state(_decoder); #endif } -bool FlacInputStream::flush() { +inline FLAC__StreamDecoderState FlacInputStream::getStreamDecoderState() const { assert(isValid()); - flushBuffer(); #ifdef LEGACY_FLAC - return 0 != ::FLAC__seekable_stream_decoder_flush(_decoder); + return ::FLAC__seekable_stream_decoder_get_stream_decoder_state(_decoder); #else - return 0 != ::FLAC__stream_decoder_flush(_decoder); + return ::FLAC__stream_decoder_get_state(_decoder); #endif } @@ -451,8 +376,6 @@ } inline ::FLAC__SeekableStreamDecoderReadStatus FlacInputStream::callbackRead(FLAC__byte buffer[], FLAC_size_t *bytes) { - assert(_fileInfo.fileHandle != NULL); - if (*bytes == 0) #ifdef LEGACY_FLAC return FLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_ERROR; /* abort to avoid a deadlock */ @@ -460,19 +383,15 @@ return FLAC__STREAM_DECODER_READ_STATUS_ABORT; /* abort to avoid a deadlock */ #endif - const uint32 length = MIN(_fileInfo.fileEndPos - _fileInfo.filePos, static_cast<uint32>(*bytes)); + const uint32 bytesRead = _inStream->read(buffer, *bytes); - _fileInfo.fileHandle->seek(_fileInfo.filePos); - const uint32 bytesRead = _fileInfo.fileHandle->read(buffer, length); - - if (bytesRead == 0 && _fileInfo.fileHandle->ioFailed()) + if (bytesRead == 0 && _inStream->ioFailed()) #ifdef LEGACY_FLAC return FLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_ERROR; #else return FLAC__STREAM_DECODER_READ_STATUS_ABORT; #endif - _fileInfo.filePos += bytesRead; *bytes = static_cast<uint>(bytesRead); #ifdef LEGACY_FLAC return FLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_OK; @@ -481,12 +400,6 @@ #endif } -inline void FlacInputStream::setLastSample(FLAC__uint64 absoluteSample) { - if (_lastSampleWritten && absoluteSample > _lastSample) - _lastSampleWritten = false; - _lastSample = absoluteSample; -} - inline void FlacInputStream::flushBuffer() { _lastSampleWritten = _lastSampleWritten && _preBuffer.bufFill == 0; _preBuffer.bufFill = 0; @@ -751,12 +664,9 @@ } inline ::FLAC__SeekableStreamDecoderSeekStatus FlacInputStream::callbackSeek(FLAC__uint64 absoluteByteOffset) { - FLAC__uint64 newPos = absoluteByteOffset + _fileInfo.fileStartPos; - const bool result = (newPos < _fileInfo.fileEndPos); + _inStream->seek(absoluteByteOffset, SEEK_SET); + const bool result = (absoluteByteOffset == _inStream->pos()); - if (result) - _fileInfo.filePos = static_cast<uint32>(newPos); - #ifdef LEGACY_FLAC return result ? FLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_OK : FLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_ERROR; #else @@ -765,9 +675,7 @@ } inline ::FLAC__SeekableStreamDecoderTellStatus FlacInputStream::callbackTell(FLAC__uint64 *absoluteByteOffset) { - /*if () - return FLAC__SEEKABLE_STREAM_DECODER_TELL_STATUS_ERROR;*/ - *absoluteByteOffset = static_cast<FLAC__uint64>(_fileInfo.filePos-_fileInfo.fileStartPos); + *absoluteByteOffset = static_cast<FLAC__uint64>(_inStream->pos()); #ifdef LEGACY_FLAC return FLAC__SEEKABLE_STREAM_DECODER_TELL_STATUS_OK; #else @@ -776,15 +684,8 @@ } inline ::FLAC__SeekableStreamDecoderLengthStatus FlacInputStream::callbackLength(FLAC__uint64 *streamLength) { - if (_fileInfo.fileStartPos > _fileInfo.fileEndPos) + *streamLength = static_cast<FLAC__uint64>(_inStream->size()); #ifdef LEGACY_FLAC - return FLAC__SEEKABLE_STREAM_DECODER_LENGTH_STATUS_ERROR; -#else - return FLAC__STREAM_DECODER_LENGTH_STATUS_ERROR; -#endif - - *streamLength = static_cast<FLAC__uint64>(_fileInfo.fileEndPos - _fileInfo.fileStartPos); -#ifdef LEGACY_FLAC return FLAC__SEEKABLE_STREAM_DECODER_LENGTH_STATUS_OK; #else return FLAC__STREAM_DECODER_LENGTH_STATUS_OK; @@ -792,7 +693,7 @@ } inline bool FlacInputStream::callbackEOF() { - return _fileInfo.filePos >= _fileInfo.fileEndPos; + return _inStream->eos(); } @@ -895,10 +796,10 @@ } // Next, try to create a FlacInputStream from it - FlacInputStream *tempStream = new FlacInputStream(&file); + FlacInputStream *tempStream = new FlacInputStream(&file, false); // If initialising the stream fails, we set the error flag - if (!tempStream || !tempStream->init()) + if (!tempStream || !tempStream->isStreamDecoderReady()) _errorFlag = true; delete tempStream; @@ -919,28 +820,19 @@ return; } - // Create an AudioStream from the file - FlacInputStream *flac = new FlacInputStream(file); - flac->init(); - file->decRef(); + // Convert startFrame & duration from frames (1/75 s) to milliseconds (1/1000s) + uint start = startFrame * 1000 / 75; + uint end = duration ? ((startFrame + duration) * 1000 / 75) : 0; - // Seek to the correct start position and start playback - if (flac->isStreamDecoderReady()) { - const FLAC__StreamMetadata_StreamInfo &info = flac->getStreamInfo(); - if (duration) - flac->setLastSample(static_cast<FLAC__uint64>(startFrame + duration) * (info.sample_rate / 75)); - else - flac->setLastSample(0); - - if (flac->seekAbsolute(static_cast<FLAC__uint64>(startFrame) * (info.sample_rate / 75))) { - mixer->playInputStream(Audio::Mixer::kMusicSoundType, handle, flac); - return; - } - // startSample is beyond the existing Samples - debug(1, "FlacTrackInfo: Audiostream %s could not seek to frame %d (ca %d secs)", _filename.c_str(), startFrame, startFrame/75); - flac->finish(); + // ... create an AudioStream ... + FlacInputStream *input = new FlacInputStream(file, true, start, end); + if (!input->isStreamDecoderReady()) { + delete input; + return; } - delete flac; + + // ... and play it + mixer->playInputStream(Audio::Mixer::kMusicSoundType, handle, input); } DigitalTrackInfo* getFlacTrack(int track) { @@ -964,19 +856,46 @@ return NULL; } -AudioStream *makeFlacStream(File *file, uint32 length) { - assert(file != NULL); - uint32 start = file->pos(); +AudioStream *makeFlacStream(File *file, uint32 size) { + assert(file); - FlacInputStream *flac = new FlacInputStream(file, start, start + length); - if (flac->init()) - return flac; + // FIXME: For now, just read the whole data into memory, and be done + // with it. Of course this is in general *not* a nice thing to do... - delete flac; - return NULL; + // If no size was specified, read the whole remainder of the file + if (!size) + size = file->size() - file->pos(); + + // Read 'size' bytes of data into a MemoryReadStream + Common::MemoryReadStream *stream = file->readStream(size); + + // .. and create an MP3InputStream from all this + FlacInputStream *input = new FlacInputStream(stream, true); + if (!input->isStreamDecoderReady()) { + delete input; + return 0; + } + return input; } +AudioStream *makeFlacStream( + Common::SeekableReadStream *stream, + bool disposeAfterUse, + uint32 startTime, + uint32 duration, + uint numLoops) { + uint32 endTime = duration ? (startTime + duration) : 0; + + FlacInputStream *input = new FlacInputStream(stream, disposeAfterUse, startTime, endTime, numLoops); + if (!input->isStreamDecoderReady()) { + delete input; + return 0; + } + return input; +} + + } // End of namespace Audio #endif // #ifdef USE_FLAC Modified: scummvm/trunk/sound/flac.h =================================================================== --- scummvm/trunk/sound/flac.h 2007-02-22 18:03:05 UTC (rev 25791) +++ scummvm/trunk/sound/flac.h 2007-02-22 18:21:01 UTC (rev 25792) @@ -30,6 +30,7 @@ namespace Common { class File; + class SeekableReadStream; } namespace Audio { @@ -47,6 +48,27 @@ */ AudioStream *makeFlacStream(Common::File *file, uint32 size); + +/** + * Create a new AudioStream from the FLAC data in the given stream. + * Allows for looping (which is why we require a SeekableReadStream), + * and specifying only a portion of the data to be played, based + * on time offsets. + * + * @param stream the SeekableReadStream from which to read the MP3 data + * @param disposeAfterUse whether to delete the stream after use + * @param startTime the (optional) time offset in milliseconds from which to start playback + * @param duration the (optional) time in milliseconds specifying how long to play + * @param numLoops how often the data shall be looped (0 = infinite) + * @return a new AudioStream, or NULL, if an error occured + */ +AudioStream *makeFlacStream( + Common::SeekableReadStream *stream, + bool disposeAfterUse, + uint32 startTime = 0, + uint32 duration = 0, + uint numLoops = 1); + } // End of namespace Audio #endif // #ifdef USE_FLAC Modified: scummvm/trunk/sound/vorbis.cpp =================================================================== --- scummvm/trunk/sound/vorbis.cpp 2007-02-22 18:03:05 UTC (rev 25791) +++ scummvm/trunk/sound/vorbis.cpp 2007-02-22 18:21:01 UTC (rev 25792) @@ -332,7 +332,6 @@ void VorbisTrackInfo::play(Audio::Mixer *mixer, Audio::SoundHandle *handle, int startFrame, int duration) { assert(!_errorFlag); - uint start, end; // Open the file Common::File *file = new Common::File(); @@ -343,8 +342,8 @@ } // Convert startFrame & duration from frames (1/75 s) to milliseconds (1/1000s) - start = startFrame * 1000 / 75; - end = duration ? ((startFrame + duration) * 1000 / 75) : 0; + uint start = startFrame * 1000 / 75; + uint end = duration ? ((startFrame + duration) * 1000 / 75) : 0; // ... create an AudioStream ... VorbisInputStream *input = new VorbisInputStream(file, true, start, end); This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.

[Scummvm-cvs-logs] SF.net SVN: scummvm: [25794] scummvm/trunk/sound From: <fingolfin@us...> - 2007-02-22 18:36 Revision: 25794 http://scummvm.svn.sourceforge.net/scummvm/?rev=25794&view=rev Author: fingolfin Date: 2007-02-22 10:35:57 -0800 (Thu, 22 Feb 2007) Log Message: ----------- cleanup Modified Paths: -------------- scummvm/trunk/sound/mp3.cpp scummvm/trunk/sound/vorbis.cpp Modified: scummvm/trunk/sound/mp3.cpp =================================================================== --- scummvm/trunk/sound/mp3.cpp 2007-02-22 18:33:01 UTC (rev 25793) +++ scummvm/trunk/sound/mp3.cpp 2007-02-22 18:35:57 UTC (rev 25794) @@ -145,7 +145,6 @@ return; do { - // If necessary, load more data if (_stream.buffer == NULL || _stream.error == MAD_ERROR_BUFLEN) { if (!readMP3Data()) { @@ -288,6 +287,11 @@ } +#pragma mark - +#pragma mark --- MP3 factory functions --- +#pragma mark - + + AudioStream *makeMP3Stream(Common::File *file, uint32 size) { assert(file); @@ -345,7 +349,6 @@ void play(Audio::Mixer *mixer, Audio::SoundHandle *handle, int startFrame, int duration); }; - MP3TrackInfo::MP3TrackInfo(const char *filename) : _filename(filename), _errorFlag(false) { @@ -416,11 +419,6 @@ return NULL; } - -// Closing note: we added File::incRef and File::decRef mainly for the sake of the input streams -// If we could but get rid of it... - - } // End of namespace Audio #endif // #ifdef USE_MAD Modified: scummvm/trunk/sound/vorbis.cpp =================================================================== --- scummvm/trunk/sound/vorbis.cpp 2007-02-22 18:33:01 UTC (rev 25793) +++ scummvm/trunk/sound/vorbis.cpp 2007-02-22 18:35:57 UTC (rev 25794) @@ -41,9 +41,6 @@ #endif -using Common::File; - - namespace Audio { // These are wrapper functions to allow using a SeekableReadStream object to @@ -260,7 +257,7 @@ _bufferEnd = (int16 *)read_pos; } -AudioStream *makeVorbisStream(File *file, uint32 size) { +AudioStream *makeVorbisStream(Common::File *file, uint32 size) { assert(file); // FIXME: For now, just read the whole data into memory, and be done @@ -290,11 +287,11 @@ } - #pragma mark - #pragma mark --- Ogg Vorbis Audio CD emulation --- #pragma mark - + class VorbisTrackInfo : public DigitalTrackInfo { private: Common::String _filename; @@ -306,7 +303,6 @@ void play(Audio::Mixer *mixer, Audio::SoundHandle *handle, int startFrame, int duration); }; - VorbisTrackInfo::VorbisTrackInfo(const char *filename) : _filename(filename), _errorFlag(false) { @@ -342,7 +338,7 @@ } // Convert startFrame & duration from frames (1/75 s) to milliseconds (1/1000s), - // i.e. multiple with a factor of 1000/75 = 40/3 + // i.e. multiply with a factor of 1000/75 = 40/3. uint start = startFrame * 40 / 3; uint end = duration ? ((startFrame + duration) * 40 / 3) : 0; @@ -371,7 +367,6 @@ } - } // End of namespace Audio #endif // #ifdef USE_VORBIS This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.

[Scummvm-cvs-logs] SF.net SVN: scummvm: [25797] scummvm/trunk/sound From: <lordhoto@us...> - 2007-02-22 20:46 Revision: 25797 http://scummvm.svn.sourceforge.net/scummvm/?rev=25797&view=rev Author: lordhoto Date: 2007-02-22 12:46:09 -0800 (Thu, 22 Feb 2007) Log Message: ----------- fixed comments. Modified Paths: -------------- scummvm/trunk/sound/flac.h scummvm/trunk/sound/vorbis.h Modified: scummvm/trunk/sound/flac.h =================================================================== --- scummvm/trunk/sound/flac.h 2007-02-22 18:42:03 UTC (rev 25796) +++ scummvm/trunk/sound/flac.h 2007-02-22 20:46:09 UTC (rev 25797) @@ -55,7 +55,7 @@ * and specifying only a portion of the data to be played, based * on time offsets. * - * @param stream the SeekableReadStream from which to read the MP3 data + * @param stream the SeekableReadStream from which to read the FLAC data * @param disposeAfterUse whether to delete the stream after use * @param startTime the (optional) time offset in milliseconds from which to start playback * @param duration the (optional) time in milliseconds specifying how long to play Modified: scummvm/trunk/sound/vorbis.h =================================================================== --- scummvm/trunk/sound/vorbis.h 2007-02-22 18:42:03 UTC (rev 25796) +++ scummvm/trunk/sound/vorbis.h 2007-02-22 20:46:09 UTC (rev 25797) @@ -55,7 +55,7 @@ * and specifying only a portion of the data to be played, based * on time offsets. * - * @param stream the SeekableReadStream from which to read the MP3 data + * @param stream the SeekableReadStream from which to read the Ogg Vorbis data * @param disposeAfterUse whether to delete the stream after use * @param startTime the (optional) time offset in milliseconds from which to start playback * @param duration the (optional) time in milliseconds specifying how long to play This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.

[Scummvm-cvs-logs] SF.net SVN: scummvm: [25906] scummvm/trunk/sound From: <fingolfin@us...> - 2007-02-28 12:55 Revision: 25906 http://scummvm.svn.sourceforge.net/scummvm/?rev=25906&view=rev Author: fingolfin Date: 2007-02-28 04:54:59 -0800 (Wed, 28 Feb 2007) Log Message: ----------- cleanup Modified Paths: -------------- scummvm/trunk/sound/audiostream.cpp scummvm/trunk/sound/audiostream.h Modified: scummvm/trunk/sound/audiostream.cpp =================================================================== --- scummvm/trunk/sound/audiostream.cpp 2007-02-28 10:44:52 UTC (rev 25905) +++ scummvm/trunk/sound/audiostream.cpp 2007-02-28 12:54:59 UTC (rev 25906) @@ -32,6 +32,15 @@ #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 { @@ -62,8 +71,7 @@ { NULL, NULL, NULL } // Terminator }; -AudioStream* AudioStream::openStreamFile(const char *filename) -{ +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 Modified: scummvm/trunk/sound/audiostream.h =================================================================== --- scummvm/trunk/sound/audiostream.h 2007-02-28 10:44:52 UTC (rev 25905) +++ scummvm/trunk/sound/audiostream.h 2007-02-28 12:54:59 UTC (rev 25906) @@ -89,27 +89,6 @@ static AudioStream* openStreamFile(const char *filename); }; -/** - * A simple AudioStream which represents a 'silent' stream, - * containing the specified number of zero samples. - */ -class ZeroInputStream : public AudioStream { -private: - int _len; -public: - ZeroInputStream(uint len) : _len(len) { } - int readBuffer(int16 *buffer, const int numSamples) { - int samples = MIN(_len, numSamples); - memset(buffer, 0, samples * 2); - _len -= samples; - return samples; - } - bool isStereo() const { return false; } - bool eos() const { return _len <= 0; } - - int getRate() const { return -1; } -}; - AudioStream *makeLinearInputStream(int rate, byte flags, const byte *ptr, uint32 len, uint loopOffset, uint loopLen); /** @@ -125,15 +104,6 @@ AppendableAudioStream *makeAppendableAudioStream(int rate, byte _flags, uint32 len); -// 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)) - - } // End of namespace Audio #endif This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.

[Scummvm-cvs-logs] SF.net SVN: scummvm: [25923] scummvm/trunk/sound From: <fingolfin@us...> - 2007-03-02 14:49 Revision: 25923 http://scummvm.svn.sourceforge.net/scummvm/?rev=25923&view=rev Author: fingolfin Date: 2007-03-02 06:49:07 -0800 (Fri, 02 Mar 2007) Log Message: ----------- Added reverseStereo parameter to Mixer::playInputStream; changed Mixer::playRaw to use Mixer::playInputStream Modified Paths: -------------- scummvm/trunk/sound/mixer.cpp scummvm/trunk/sound/mixer.h Modified: scummvm/trunk/sound/mixer.cpp =================================================================== --- scummvm/trunk/sound/mixer.cpp 2007-03-02 12:57:38 UTC (rev 25922) +++ scummvm/trunk/sound/mixer.cpp 2007-03-02 14:49:07 UTC (rev 25923) @@ -155,18 +155,7 @@ uint32 size, uint rate, byte flags, int id, byte volume, int8 balance, uint32 loopStart, uint32 loopEnd) { - Common::StackLock lock(_mutex); - // Prevent duplicate sounds - if (id != -1) { - for (int i = 0; i != NUM_CHANNELS; i++) - if (_channels[i] != 0 && _channels[i]->getId() == id) { - if ((flags & Mixer::FLAG_AUTOFREE) != 0) - free(sound); - return; - } - } - // Create the input stream AudioStream *input; if (flags & Mixer::FLAG_LOOP) { @@ -179,12 +168,9 @@ } else { input = makeLinearInputStream(rate, flags, (byte *)sound, size, 0, 0); } - - // Create the channel - Channel *chan = new Channel(this, type, input, true, (flags & Mixer::FLAG_REVERSE_STEREO) != 0, id); - chan->setVolume(volume); - chan->setBalance(balance); - insertChannel(handle, chan); + + // Play it + playInputStream(type, handle, input, id, volume, balance, true, false, (flags & Mixer::FLAG_REVERSE_STEREO)); } void Mixer::playInputStream( @@ -193,7 +179,8 @@ AudioStream *input, int id, byte volume, int8 balance, bool autofreeStream, - bool permanent) { + bool permanent, + bool reverseStereo) { Common::StackLock lock(_mutex); if (input == 0) { @@ -212,7 +199,7 @@ } // Create the channel - Channel *chan = new Channel(this, type, input, autofreeStream, false, id, permanent); + Channel *chan = new Channel(this, type, input, autofreeStream, reverseStereo, id, permanent); chan->setVolume(volume); chan->setBalance(balance); insertChannel(handle, chan); Modified: scummvm/trunk/sound/mixer.h =================================================================== --- scummvm/trunk/sound/mixer.h 2007-03-02 12:57:38 UTC (rev 25922) +++ scummvm/trunk/sound/mixer.h 2007-03-02 14:49:07 UTC (rev 25923) @@ -158,13 +158,16 @@ * freed after playback finished * @param permanent a flag indicating whether a plain stopAll call should * not stop this particular stream + * @param reverseStereo a flag indicating whether left and right channels shall be swapped */ void playInputStream( SoundType type, SoundHandle *handle, AudioStream *input, int id = -1, byte volume = 255, int8 balance = 0, - bool autofreeStream = true, bool permanent = false); + bool autofreeStream = true, + bool permanent = false, + bool reverseStereo = false); This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.

[Scummvm-cvs-logs] SF.net SVN: scummvm: [26428] scummvm/trunk/sound From: <eriktorbjorn@us...> - 2007-04-09 10:53 Revision: 26428 http://scummvm.svn.sourceforge.net/scummvm/?rev=26428&view=rev Author: eriktorbjorn Date: 2007-04-09 03:53:17 -0700 (Mon, 09 Apr 2007) Log Message: ----------- I hope this fixes the file headers. Modified Paths: -------------- scummvm/trunk/sound/aiff.cpp scummvm/trunk/sound/aiff.h Modified: scummvm/trunk/sound/aiff.cpp =================================================================== --- scummvm/trunk/sound/aiff.cpp 2007-04-09 10:47:26 UTC (rev 26427) +++ scummvm/trunk/sound/aiff.cpp 2007-04-09 10:53:17 UTC (rev 26428) @@ -16,7 +16,7 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA. * * $URL$ - * $Id: wave.cpp 25926 2007-03-02 15:26:22Z fingolfin $ + * $Id$ * */ Modified: scummvm/trunk/sound/aiff.h =================================================================== --- scummvm/trunk/sound/aiff.h 2007-04-09 10:47:26 UTC (rev 26427) +++ scummvm/trunk/sound/aiff.h 2007-04-09 10:53:17 UTC (rev 26428) @@ -16,7 +16,7 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA. * * $URL$ - * $Id: wave.h 22231 2006-04-29 22:33:31Z fingolfin $ + * $Id$ * */ This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.

[Scummvm-cvs-logs] SF.net SVN: scummvm: [26429] scummvm/trunk/sound From: <eriktorbjorn@us...> - 2007-04-09 10:58 Revision: 26429 http://scummvm.svn.sourceforge.net/scummvm/?rev=26429&view=rev Author: eriktorbjorn Date: 2007-04-09 03:58:14 -0700 (Mon, 09 Apr 2007) Log Message: ----------- Let's see if this works... Property Changed: ---------------- scummvm/trunk/sound/aiff.cpp scummvm/trunk/sound/aiff.h Property changes on: scummvm/trunk/sound/aiff.cpp ___________________________________________________________________ Name: svn:keywords - "Date Rev Author URL Id" + Date Rev Author URL Id Property changes on: scummvm/trunk/sound/aiff.h ___________________________________________________________________ Name: svn:keywords - "Date Rev Author URL Id" + Date Rev Author URL Id This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.

[Scummvm-cvs-logs] SF.net SVN: scummvm: [26471] scummvm/trunk/sound From: <fingolfin@us...> - 2007-04-14 17:12 Revision: 26471 http://scummvm.svn.sourceforge.net/scummvm/?rev=26471&view=rev Author: fingolfin Date: 2007-04-14 10:12:43 -0700 (Sat, 14 Apr 2007) Log Message: ----------- Extended Audio::openStreamFile function with startTime, duration and numLoops parameters, and slightly cleaned up its code Modified Paths: -------------- scummvm/trunk/sound/audiostream.cpp scummvm/trunk/sound/audiostream.h Modified: scummvm/trunk/sound/audiostream.cpp =================================================================== --- scummvm/trunk/sound/audiostream.cpp 2007-04-14 15:13:45 UTC (rev 26470) +++ scummvm/trunk/sound/audiostream.cpp 2007-04-14 17:12:43 UTC (rev 26471) @@ -59,36 +59,29 @@ static const StreamFileFormat STREAM_FILEFORMATS[] = { /* decoderName, fileExt, openStreamFuntion */ #ifdef USE_FLAC - { "Flac", "flac", makeFlacStream }, - { "Flac", "fla", makeFlacStream }, + { "Flac", ".flac", makeFlacStream }, + { "Flac", ".fla", makeFlacStream }, #endif #ifdef USE_VORBIS - { "Ogg Vorbis", "ogg", makeVorbisStream }, + { "Ogg Vorbis", ".ogg", makeVorbisStream }, #endif #ifdef USE_MAD - { "MPEG Layer 3", "mp3", makeMP3Stream }, + { "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* AudioStream::openStreamFile(const Common::String &basename, uint32 startTime, uint32 duration, uint numLoops) { 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); + Common::String filename = basename + STREAM_FILEFORMATS[i].fileExtension; + fileHandle->open(filename); if (fileHandle->isOpen()) { - stream = STREAM_FILEFORMATS[i].openStreamFile(fileHandle, true, 0, 0, 1); + // Create the stream object + stream = STREAM_FILEFORMATS[i].openStreamFile(fileHandle, true, startTime, duration, numLoops); fileHandle = 0; break; } @@ -97,7 +90,7 @@ delete fileHandle; if (stream == NULL) { - debug(1, "AudioStream: Could not open compressed AudioFile %s", filename); + debug(1, "AudioStream: Could not open compressed AudioFile %s", basename.c_str()); } return stream; Modified: scummvm/trunk/sound/audiostream.h =================================================================== --- scummvm/trunk/sound/audiostream.h 2007-04-14 15:13:45 UTC (rev 26470) +++ scummvm/trunk/sound/audiostream.h 2007-04-14 17:12:43 UTC (rev 26471) @@ -83,11 +83,14 @@ * Tries to load a file by trying all available formats. * In case of an error, the file handle will be closed, but deleting * it is still the responsibilty of the caller. - * @param filename a filename without an extension + * @param basename a filename without an extension + * @param startTime the (optional) time offset in milliseconds from which to start playback + * @param duration the (optional) time in milliseconds specifying how long to play + * @param numLoops how often the data shall be looped (0 = infinite) * @return an Audiostream ready to use in case of success; * NULL in case of an error (e.g. invalid/nonexisting file) */ - static AudioStream* openStreamFile(const char *filename); + static AudioStream* openStreamFile(const Common::String &basename, uint32 startTime = 0, uint32 duration = 0, uint numLoops = 1); }; /** This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.

[Scummvm-cvs-logs] SF.net SVN: scummvm: [26476] scummvm/trunk/sound From: <fingolfin@us...> - 2007-04-14 18:48 Revision: 26476 http://scummvm.svn.sourceforge.net/scummvm/?rev=26476&view=rev Author: fingolfin Date: 2007-04-14 11:48:09 -0700 (Sat, 14 Apr 2007) Log Message: ----------- Rewrote AudioCDManager to not use DigitalTrackInfo instances anymore, but rather create/use track AudioStream instances directly Modified Paths: -------------- scummvm/trunk/sound/audiocd.cpp scummvm/trunk/sound/audiocd.h Modified: scummvm/trunk/sound/audiocd.cpp =================================================================== --- scummvm/trunk/sound/audiocd.cpp 2007-04-14 18:44:15 UTC (rev 26475) +++ scummvm/trunk/sound/audiocd.cpp 2007-04-14 18:48:09 UTC (rev 26476) @@ -23,6 +23,7 @@ #include "common/stdafx.h" #include "sound/audiocd.h" +#include "sound/audiostream.h" #include "sound/mp3.h" #include "sound/vorbis.h" #include "sound/flac.h" @@ -35,87 +36,74 @@ namespace Audio { -struct TrackFormat { - /** Decodername */ - const char* decoderName; - /** - * Pointer to a function which tries to open the specified track - the only argument - * is the number of the track to be played. - * Returns either a DigitalTrackInfo object representing the requested track or null - * in case of an error - */ - DigitalTrackInfo* (*openTrackFunction)(int); -}; - -static const TrackFormat s_trackFormats[] = { - /* decoderName, openTrackFunction */ -#ifdef USE_FLAC - { "Flac", getFlacTrack }, -#endif -#ifdef USE_VORBIS - { "Ogg Vorbis", getVorbisTrack }, -#endif -#ifdef USE_MAD - { "MPEG Layer 3", getMP3Track }, -#endif - - { NULL, NULL } // Terminator -}; - - AudioCDManager::AudioCDManager() { - memset(_cachedTracks, 0, sizeof(_cachedTracks)); - memset(_trackInfo, 0, sizeof(_trackInfo)); _cd.playing = false; _cd.track = 0; _cd.start = 0; _cd.duration = 0; _cd.numLoops = 0; - _currentCacheIdx = 0; _mixer = g_system->getMixer(); assert(_mixer); } void AudioCDManager::play(int track, int numLoops, int startFrame, int duration) { if (numLoops != 0 || startFrame != 0) { - // Try to load the track from a .mp3/.ogg file, and if found, use - // that. If not found, attempt to do regular Audio CD playback of - // the requested track. - int index = getCachedTrack(track); - _cd.track = track; _cd.numLoops = numLoops; _cd.start = startFrame; _cd.duration = duration; - if (index >= 0) { - _mixer->stopHandle(_cd.handle); - _cd.playing = true; + // Try to load the track from a compressed data file, and if found, use + // that. If not found, attempt to start regular Audio CD playback of + // the requested track. + char trackName[2][16]; + sprintf(trackName[0], "track%d", track); + sprintf(trackName[1], "track%02d", track); + Audio::AudioStream *stream = 0; + + for (int i = 0; !stream && i < 2; ++i) { /* FIXME: Seems numLoops == 0 and numLoops == 1 both indicate a single repetition, while all other positive numbers indicate precisely the number of desired repetitions. Finally, -1 means infinitely many */ - numLoops = (numLoops < 1) ? numLoops + 1 : numLoops; - _trackInfo[index]->play(_mixer, &_cd.handle, numLoops, _cd.start, _cd.duration); + // We multiply by 40 / 3 = 1000 / 75 to convert frames to milliseconds + stream = AudioStream::openStreamFile(trackName[i], startFrame * 40 / 3, duration * 40 / 3, (numLoops < 1) ? numLoops + 1 : numLoops); + } + + // Stop any currently playing emulated track + _mixer->stopHandle(_cd.handle); + + // HACK: We abuse _cd.playing to store whether we are playing a real or an emulated track. + if (stream != 0) { + _cd.playing = true; + _mixer->playInputStream(Audio::Mixer::kMusicSoundType, &_cd.handle, stream); } else { - g_system->playCD(track, numLoops, startFrame, duration); _cd.playing = false; + g_system->playCD(track, numLoops, startFrame, duration); } } } void AudioCDManager::stop() { if (_cd.playing) { + // Audio CD emulation _mixer->stopHandle(_cd.handle); _cd.playing = false; } else { + // Real Audio CD g_system->stopCD(); } } bool AudioCDManager::isPlaying() const { - return _cd.playing || g_system->pollCD(); + if (_cd.playing) { + // Audio CD emulation + return _mixer->isSoundHandleActive(_cd.handle); + } else { + // Real Audio CD + return g_system->pollCD(); + } } void AudioCDManager::updateCD() { @@ -125,7 +113,8 @@ // FIXME: We do not update the numLoops parameter here (and in fact, // currently can't do that). Luckily, only one engine ever checks // this part of the AudioCD status, namely the SCUMM engine; and it - // only checks + // only checks whether the track is currently set to infinite looping + // or not. _cd.playing = false; } } else { @@ -141,41 +130,4 @@ return info; } -int AudioCDManager::getCachedTrack(int track) { - // See if we find the track in the cache - for (int i = 0; i < CACHE_TRACKS; i++) - if (_cachedTracks[i] == track) { - return _trackInfo[i] ? i : -1; - } - - // The track is not already in the cache. Try and see if - // we can load it. - DigitalTrackInfo *newTrack = 0; - for (const TrackFormat *format = s_trackFormats; - format->openTrackFunction != NULL && newTrack == NULL; - ++format) { - newTrack = format->openTrackFunction(track); - } - - int currentIndex = -1; - - if (newTrack != NULL) { - // We successfully loaded a digital track. Store it into _trackInfo. - - currentIndex = _currentCacheIdx++; - _currentCacheIdx %= CACHE_TRACKS; - - // First, delete the previous track info object - delete _trackInfo[currentIndex]; - - // Then, store the new track info object - _trackInfo[currentIndex] = newTrack; - _cachedTracks[currentIndex] = track; - } else { - debug(2, "Track %d not available in compressed format", track); - } - - return currentIndex; -} - } // End of namespace Audio Modified: scummvm/trunk/sound/audiocd.h =================================================================== --- scummvm/trunk/sound/audiocd.h 2007-04-14 18:44:15 UTC (rev 26475) +++ scummvm/trunk/sound/audiocd.h 2007-04-14 18:48:09 UTC (rev 26476) @@ -73,27 +73,12 @@ friend class Common::Singleton<SingletonBaseType>; AudioCDManager(); - int getCachedTrack(int track); - -private: /* used for emulated CD music */ struct ExtStatus : Status { SoundHandle handle; }; ExtStatus _cd; - enum { -#if defined(__PSP__) - CACHE_TRACKS = 4 //the PSP can't have more than 8 files open simultaneously - //so don't use more than 4 filehandles for CD tracks -#else - CACHE_TRACKS = 10 -#endif - }; - int _cachedTracks[CACHE_TRACKS]; - DigitalTrackInfo *_trackInfo[CACHE_TRACKS]; - int _currentCacheIdx; - Mixer *_mixer; }; This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.

[Scummvm-cvs-logs] SF.net SVN: scummvm: [26477] scummvm/trunk/sound From: <fingolfin@us...> - 2007-04-14 18:51 Revision: 26477 http://scummvm.svn.sourceforge.net/scummvm/?rev=26477&view=rev Author: fingolfin Date: 2007-04-14 11:51:38 -0700 (Sat, 14 Apr 2007) Log Message: ----------- Got rid of the last traces of DigitalTrackInfo Modified Paths: -------------- scummvm/trunk/sound/audiocd.h scummvm/trunk/sound/flac.cpp scummvm/trunk/sound/flac.h scummvm/trunk/sound/mp3.cpp scummvm/trunk/sound/mp3.h scummvm/trunk/sound/vorbis.cpp scummvm/trunk/sound/vorbis.h Modified: scummvm/trunk/sound/audiocd.h =================================================================== --- scummvm/trunk/sound/audiocd.h 2007-04-14 18:48:09 UTC (rev 26476) +++ scummvm/trunk/sound/audiocd.h 2007-04-14 18:51:38 UTC (rev 26477) @@ -32,15 +32,6 @@ namespace Audio { -class DigitalTrackInfo { -public: - virtual ~DigitalTrackInfo() {} - - virtual void play(Mixer *mixer, SoundHandle *handle, int numLoops, int startFrame, int duration) = 0; -// virtual void stop(); -}; - - class AudioCDManager : public Common::Singleton<AudioCDManager> { public: struct Status { Modified: scummvm/trunk/sound/flac.cpp =================================================================== --- scummvm/trunk/sound/flac.cpp 2007-04-14 18:48:09 UTC (rev 26476) +++ scummvm/trunk/sound/flac.cpp 2007-04-14 18:51:38 UTC (rev 26477) @@ -770,96 +770,6 @@ return input; } - -#pragma mark - -#pragma mark --- Flac Audio CD emulation --- -#pragma mark - - - -class FlacTrackInfo : public DigitalTrackInfo { -private: - Common::String _filename; - bool _errorFlag; - -public: - FlacTrackInfo(const char *filename); - bool error() { return _errorFlag; } - void play(Mixer *mixer, SoundHandle *handle, int numLoops, int startFrame, int duration); -}; - -FlacTrackInfo::FlacTrackInfo(const char *filename) : - _filename(filename), - _errorFlag(false) { - - // Try to open the file - Common::File file; - if (!file.open(_filename)) { - _errorFlag = true; - return; - } - - // Next, try to create a FlacInputStream from it - FlacInputStream *tempStream = new FlacInputStream(&file, false); - - // If initialising the stream fails, we set the error flag - if (!tempStream || !tempStream->isStreamDecoderReady()) - _errorFlag = true; - - delete tempStream; -} - -void FlacTrackInfo::play(Mixer *mixer, SoundHandle *handle, int numLoops, int startFrame, int duration) { - assert(!_errorFlag); - - if (error()) { - debug(1, "FlacTrackInfo::play: invalid state, method should not been called"); - } - - // Open the file - Common::File *file = new Common::File(); - if (!file || !file->open(_filename)) { - warning("FlacTrackInfo::play: failed to open '%s'", _filename.c_str()); - delete file; - return; - } - - // Convert startFrame & duration from frames (1/75 s) to milliseconds (1/1000s) - uint start = startFrame * 1000 / 75; - uint end = duration ? ((startFrame + duration) * 1000 / 75) : 0; - - // ... create an AudioStream ... - FlacInputStream *input = new FlacInputStream(file, true, start, end, numLoops); - if (!input->isStreamDecoderReady()) { - delete input; - return; - } - - // ... and play it - mixer->playInputStream(Audio::Mixer::kMusicSoundType, handle, input); -} - -DigitalTrackInfo* getFlacTrack(int track) { - assert(track >= 1); - char trackName[4][32]; - - sprintf(trackName[0], "track%d.flac", track); - sprintf(trackName[1], "track%02d.flac", track); - sprintf(trackName[2], "track%d.fla", track); - sprintf(trackName[3], "track%02d.fla", track); - - for (int i = 0; i < 4; ++i) { - if (Common::File::exists(trackName[i])) { - FlacTrackInfo *trackInfo = new FlacTrackInfo(trackName[i]); - if (!trackInfo->error()) - return trackInfo; - delete trackInfo; - } - } - - return NULL; -} - - } // End of namespace Audio #endif // #ifdef USE_FLAC Modified: scummvm/trunk/sound/flac.h =================================================================== --- scummvm/trunk/sound/flac.h 2007-04-14 18:48:09 UTC (rev 26476) +++ scummvm/trunk/sound/flac.h 2007-04-14 18:51:38 UTC (rev 26477) @@ -36,10 +36,7 @@ namespace Audio { class AudioStream; -class DigitalTrackInfo; -DigitalTrackInfo *getFlacTrack(int track); - /** * Create a new AudioStream from the FLAC data in the given * file. If you only want to play part of that file, then seek Modified: scummvm/trunk/sound/mp3.cpp =================================================================== --- scummvm/trunk/sound/mp3.cpp 2007-04-14 18:48:09 UTC (rev 26476) +++ scummvm/trunk/sound/mp3.cpp 2007-04-14 18:51:38 UTC (rev 26477) @@ -339,93 +339,6 @@ return new MP3InputStream(stream, disposeAfterUse, start, end, numLoops); } - -#pragma mark - -#pragma mark --- MP3 Audio CD emulation --- -#pragma mark - - - -class MP3TrackInfo : public DigitalTrackInfo { -private: - Common::String _filename; - bool _errorFlag; - -public: - MP3TrackInfo(const char *filename); - bool error() { return _errorFlag; } - void play(Mixer *mixer, SoundHandle *handle, int numLoops, int startFrame, int duration); -}; - -MP3TrackInfo::MP3TrackInfo(const char *filename) : - _filename(filename), - _errorFlag(false) { - - // Try to open the file - Common::File file; - if (!file.open(_filename)) { - _errorFlag = true; - return; - } - - // Next, try to create an MP3InputStream from it - MP3InputStream *tempStream = new MP3InputStream(&file, false); - - // If we see EOS here then that means that not (enough) valid input - // data was given. - _errorFlag = tempStream->endOfData(); - - // Clean up again - delete tempStream; -} - -void MP3TrackInfo::play(Mixer *mixer, SoundHandle *handle, int numLoops, int startFrame, int duration) { - assert(!_errorFlag); - - mad_timer_t start; - mad_timer_t end; - - // Both startFrame and duration are given in frames, where 75 frames are one second. - // Calculate the appropriate mad_timer_t values from them. - mad_timer_set(&start, startFrame / 75, startFrame % 75, 75); - if (duration == 0) { - end = mad_timer_zero; - } else { - int endFrame = startFrame + duration; - mad_timer_set(&end, endFrame / 75, endFrame % 75, 75); - } - - // Open the file - Common::File *file = new Common::File(); - if (!file || !file->open(_filename)) { - warning("MP3TrackInfo::play: failed to open '%s'", _filename.c_str()); - delete file; - return; - } - - // ... create an AudioStream ... - MP3InputStream *input = new MP3InputStream(file, true, start, end, numLoops); - - // ... and play it - mixer->playInputStream(Audio::Mixer::kMusicSoundType, handle, input); -} - -DigitalTrackInfo *getMP3Track(int track) { - char trackName[2][32]; - - sprintf(trackName[0], "track%d.mp3", track); - sprintf(trackName[1], "track%02d.mp3", track); - - for (int i = 0; i < 2; ++i) { - if (Common::File::exists(trackName[i])) { - MP3TrackInfo *trackInfo = new MP3TrackInfo(trackName[i]); - if (!trackInfo->error()) - return trackInfo; - delete trackInfo; - } - } - return NULL; -} - } // End of namespace Audio #endif // #ifdef USE_MAD Modified: scummvm/trunk/sound/mp3.h =================================================================== --- scummvm/trunk/sound/mp3.h 2007-04-14 18:48:09 UTC (rev 26476) +++ scummvm/trunk/sound/mp3.h 2007-04-14 18:51:38 UTC (rev 26477) @@ -36,10 +36,7 @@ namespace Audio { class AudioStream; -class DigitalTrackInfo; -DigitalTrackInfo *getMP3Track(int track); - /** * Create a new AudioStream from the MP3 data in the given * file. If you only want to play part of that file, then seek Modified: scummvm/trunk/sound/vorbis.cpp =================================================================== --- scummvm/trunk/sound/vorbis.cpp 2007-04-14 18:48:09 UTC (rev 26476) +++ scummvm/trunk/sound/vorbis.cpp 2007-04-14 18:51:38 UTC (rev 26477) @@ -293,86 +293,6 @@ } -#pragma mark - -#pragma mark --- Ogg Vorbis Audio CD emulation --- -#pragma mark - - - -class VorbisTrackInfo : public DigitalTrackInfo { -private: - Common::String _filename; - bool _errorFlag; - -public: - VorbisTrackInfo(const char *filename); - bool error() { return _errorFlag; } - void play(Mixer *mixer, SoundHandle *handle, int numLoops, int startFrame, int duration); -}; - -VorbisTrackInfo::VorbisTrackInfo(const char *filename) : - _filename(filename), - _errorFlag(false) { - - - // Try to open the file - Common::File file; - if (!file.open(_filename)) { - _errorFlag = true; - return; - } - - // Next, try to create a VorbisInputStream from it - VorbisInputStream *tempStream = new VorbisInputStream(&file, false); - - // If an error occured... - // TODO: add an error or init method to VorbisInputStream - _errorFlag = tempStream->endOfData(); - - // Clean up again - delete tempStream; -} - -void VorbisTrackInfo::play(Mixer *mixer, SoundHandle *handle, int numLoops, int startFrame, int duration) { - assert(!_errorFlag); - - // Open the file - Common::File *file = new Common::File(); - if (!file || !file->open(_filename)) { - warning("VorbisTrackInfo::play: failed to open '%s'", _filename.c_str()); - delete file; - return; - } - - // Convert startFrame & duration from frames (1/75 s) to milliseconds (1/1000s), - // i.e. multiply with a factor of 1000/75 = 40/3. - uint start = startFrame * 40 / 3; - uint end = duration ? ((startFrame + duration) * 40 / 3) : 0; - - // ... create an AudioStream ... - VorbisInputStream *input = new VorbisInputStream(file, true, start, end, numLoops); - - // ... and play it - mixer->playInputStream(Audio::Mixer::kMusicSoundType, handle, input); -} - -DigitalTrackInfo *getVorbisTrack(int track) { - char trackName[2][32]; - - sprintf(trackName[0], "track%d.ogg", track); - sprintf(trackName[1], "track%02d.ogg", track); - - for (int i = 0; i < 2; ++i) { - if (Common::File::exists(trackName[i])) { - VorbisTrackInfo *trackInfo = new VorbisTrackInfo(trackName[i]); - if (!trackInfo->error()) - return trackInfo; - delete trackInfo; - } - } - return NULL; -} - - } // End of namespace Audio #endif // #ifdef USE_VORBIS Modified: scummvm/trunk/sound/vorbis.h =================================================================== --- scummvm/trunk/sound/vorbis.h 2007-04-14 18:48:09 UTC (rev 26476) +++ scummvm/trunk/sound/vorbis.h 2007-04-14 18:51:38 UTC (rev 26477) @@ -36,10 +36,7 @@ namespace Audio { class AudioStream; -class DigitalTrackInfo; -DigitalTrackInfo *getVorbisTrack(int track); - /** * Create a new AudioStream from the Ogg Vorbis data in the given * file. If you only want to play part of that file, then seek This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.

[Scummvm-cvs-logs] SF.net SVN: scummvm: [26869] scummvm/trunk/sound From: <cyx@us...> - 2007-05-19 12:06 Revision: 26869 http://scummvm.svn.sourceforge.net/scummvm/?rev=26869&view=rev Author: cyx Date: 2007-05-19 05:06:51 -0700 (Sat, 19 May 2007) Log Message: ----------- added Amiga SoundFx module player code, used by Amiga versions of the following Delphine Software games : Future Wars, Operation Stealth, Cruise for a Corpse and Another World Modified Paths: -------------- scummvm/trunk/sound/mods/paula.h scummvm/trunk/sound/module.mk Added Paths: ----------- scummvm/trunk/sound/mods/soundfx.cpp scummvm/trunk/sound/mods/soundfx.h Modified: scummvm/trunk/sound/mods/paula.h =================================================================== --- scummvm/trunk/sound/mods/paula.h 2007-05-18 21:12:23 UTC (rev 26868) +++ scummvm/trunk/sound/mods/paula.h 2007-05-19 12:06:51 UTC (rev 26869) @@ -41,7 +41,7 @@ ~Paula(); bool playing() const { return _playing; } - void setInterruptFreq(int freq) { _intFreq = freq; } + void setInterruptFreq(int freq) { _curInt = _intFreq = freq; } void setPanning(byte voice, byte panning) { assert(voice < NUM_VOICES); _voice[voice].panning = panning; Added: scummvm/trunk/sound/mods/soundfx.cpp =================================================================== --- scummvm/trunk/sound/mods/soundfx.cpp (rev 0) +++ scummvm/trunk/sound/mods/soundfx.cpp 2007-05-19 12:06:51 UTC (rev 26869) @@ -0,0 +1,298 @@ +/* ScummVM - Scumm Interpreter + * Copyright (C) 2007 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 "sound/mods/paula.h" +#include "sound/mods/soundfx.h" +#include "sound/audiostream.h" + +namespace Audio { + +struct SoundFxInstrument { + uint8 name[23]; + uint16 len; + uint8 finetune; + uint8 volume; + uint16 repeatPos; + uint16 repeatLen; + int8 *data; +}; + +class SoundFx : public Paula { +public: + + enum { + NUM_CHANNELS = 4, + NUM_INSTRUMENTS = 15, + CIA_FREQ = 715909 + }; + + SoundFx(int rate, bool stereo); + virtual ~SoundFx(); + + bool load(Common::SeekableReadStream *data); + void play(); + +protected: + + void handlePattern(int ch, uint32 pat); + void updateEffects(int ch); + void handleTick(); + + void startPaula(); + void stopPaula(); + void setPaulaChannelPeriod(uint8 channel, int16 period); + void setPaulaChannelVolume(uint8 channel, uint8 volume); + void enablePaulaChannel(uint8 channel); + void disablePaulaChannel(uint8 channel); + void setupPaulaChannel(uint8 channel, const int8 *data, uint16 len, uint16 repeatPos, uint16 repeatLen); + + virtual void interrupt(); + + uint8 _ticks; + uint16 _delay; + SoundFxInstrument _instruments[NUM_INSTRUMENTS]; + uint8 _numOrders; + uint8 _curOrder; + uint16 _curPos; + uint8 _ordersTable[128]; + uint8 *_patternData; + int8 *_instrumentData; + int _eventsFreq; + uint16 _effects[NUM_CHANNELS]; +}; + +SoundFx::SoundFx(int rate, bool stereo) + : Paula(stereo, rate) { + _ticks = 0; + _delay = 0; + memset(_instruments, 0, sizeof(_instruments)); + _numOrders = 0; + _curOrder = 0; + _curPos = 0; + memset(_ordersTable, 0, sizeof(_ordersTable)); + _patternData = 0; + _instrumentData = 0; + _eventsFreq = 0; + memset(_effects, 0, sizeof(_effects)); +} + +SoundFx::~SoundFx() { + free(_patternData); + free(_instrumentData); +} + +bool SoundFx::load(Common::SeekableReadStream *data) { + bool loaded = false; + int instrumentsSize[15]; + int totalInstrumentsSize = 0; + for (int i = 0; i < NUM_INSTRUMENTS; ++i) { + instrumentsSize[i] = data->readUint32BE(); + totalInstrumentsSize += instrumentsSize[i]; + } + uint8 tag[4]; + data->read(tag, 4); + if (memcmp(tag, "SONG", 4) == 0) { + _delay = data->readUint16BE(); + data->skip(7 * 2); + for (int i = 0; i < NUM_INSTRUMENTS; ++i) { + SoundFxInstrument *ins = &_instruments[i]; + data->read(ins->name, 22); ins->name[22] = 0; + ins->len = data->readUint16BE(); + ins->finetune = data->readByte(); + ins->volume = data->readByte(); + ins->repeatPos = data->readUint16BE(); + ins->repeatLen = data->readUint16BE(); + } + _numOrders = data->readByte(); + data->skip(1); + data->read(_ordersTable, 128); + int maxOrder = 0; + for (int i = 0; i < _numOrders; ++i) { + if (_ordersTable[i] > maxOrder) { + maxOrder = _ordersTable[i]; + } + } + int patternSize = (maxOrder + 1) * 4 * 4 * 64; + _patternData = (uint8 *)malloc(patternSize); + if (_patternData) { + data->read(_patternData, patternSize); + _instrumentData = (int8 *)malloc(totalInstrumentsSize); + if (_instrumentData) { + data->read(_instrumentData, totalInstrumentsSize); + int8 *p = _instrumentData; + for (int i = 0; i < NUM_INSTRUMENTS; ++i) { + SoundFxInstrument *ins = &_instruments[i]; + assert(ins->len <= 2 || ins->len * 2 <= instrumentsSize[i]); + assert(ins->repeatLen <= 2 || (ins->repeatPos + ins->repeatLen) * 2 <= instrumentsSize[i]); + if (instrumentsSize[i] != 0) { + ins->data = p; + p += instrumentsSize[i]; + } + } + loaded = true; + } + } + } + return loaded; +} + +void SoundFx::play() { + _curPos = 0; + _curOrder = 0; + _ticks = 0; + _eventsFreq = CIA_FREQ / _delay; + setInterruptFreq(_rate / _eventsFreq); + startPaula(); +} + +void SoundFx::handlePattern(int ch, uint32 pat) { + uint16 note1 = pat >> 16; + uint16 note2 = pat & 0xFFFF; + if (note1 != 0xFFFD) { + int ins = (note2 & 0xF000) >> 12; + if (ins != 0) { + SoundFxInstrument *i = &_instruments[ins - 1]; + setupPaulaChannel(ch, i->data, i->len, i->repeatPos, i->repeatLen); + int effect = (note2 & 0xF00) >> 8; + int volume = i->volume; + switch (effect) { + case 5: // volume up + volume += (note2 & 0xFF); + if (volume > 63) { + volume = 63; + } + break; + case 6: // volume down + volume -= (note2 & 0xFF); + if (volume < 0) { + volume = 0; + } + break; + } + setPaulaChannelVolume(ch, volume); + } + } + _effects[ch] = note2; + if (note1 == 0xFFFD) { // PIC + _effects[ch] = 0; + } else if (note1 == 0xFFFE) { // STP + disablePaulaChannel(ch); + } else if (note1 != 0) { + setPaulaChannelPeriod(ch, note1); + enablePaulaChannel(ch); + } +} + +void SoundFx::updateEffects(int ch) { + // updateEffects() is a no-op in all Delphine Software games using SoundFx : FW,OS,Cruise,AW + if (_effects[ch] != 0) { + switch (_effects[ch]) { + case 1: // appreggiato + case 2: // pitchbend + case 3: // ledon, enable low-pass filter + case 4: // ledoff, disable low-pass filter + case 7: // set step up + case 8: // set step down + warning("Unhandled effect %d\n", _effects[ch]); + break; + } + } +} + +void SoundFx::handleTick() { + ++_ticks; + if (_ticks != 6) { + for (int ch = 0; ch < 4; ++ch) { + updateEffects(ch); + } + } else { + _ticks = 0; + const uint8 *patternData = _patternData + _ordersTable[_curOrder] * 1024 + _curPos; + for (int ch = 0; ch < 4; ++ch) { + handlePattern(ch, READ_BE_UINT32(patternData)); + patternData += 4; + } + _curPos += 4 * 4; + if (_curPos >= 1024) { + _curPos = 0; + ++_curOrder; + if (_curOrder == _numOrders) { + stopPaula(); + } + } + } +} + +void SoundFx::startPaula() { + _playing = true; + _end = false; +} + +void SoundFx::stopPaula() { + _playing = false; + _end = true; +} + +void SoundFx::setPaulaChannelPeriod(uint8 channel, int16 period) { + _voice[channel].period = period; +} + +void SoundFx::setPaulaChannelVolume(uint8 channel, uint8 volume) { + _voice[channel].volume = volume; +} + +void SoundFx::enablePaulaChannel(uint8 channel) { +} + +void SoundFx::disablePaulaChannel(uint8 channel) { + _voice[channel].period = 0; +} + +void SoundFx::setupPaulaChannel(uint8 channel, const int8 *data, uint16 len, uint16 repeatPos, uint16 repeatLen) { + if (data && len > 1) { + Channel *ch = &_voice[channel]; + ch->data = data; + ch->dataRepeat = data + repeatPos * 2; + ch->length = len * 2; + ch->lengthRepeat = repeatLen * 2; + ch->offset = 0; + } +} + +void SoundFx::interrupt() { + handleTick(); +} + +AudioStream *makeSoundFxStream(Common::SeekableReadStream *data, int rate, bool stereo) { + SoundFx *stream = new SoundFx(rate, stereo); + if (stream->load(data)) { + stream->play(); + return stream; + } + delete stream; + return 0; +} + +} // End of namespace Audio Property changes on: scummvm/trunk/sound/mods/soundfx.cpp ___________________________________________________________________ Name: svn:mime-type + text/plain Name: svn:keywords + Date Rev Author URL Id Name: svn:eol-style + native Added: scummvm/trunk/sound/mods/soundfx.h =================================================================== --- scummvm/trunk/sound/mods/soundfx.h (rev 0) +++ scummvm/trunk/sound/mods/soundfx.h 2007-05-19 12:06:51 UTC (rev 26869) @@ -0,0 +1,37 @@ + +/* ScummVM - Scumm Interpreter + * Copyright (C) 2007 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. + * + * $URL$ + * $Id$ + * + */ + +#ifndef SOUND_MODS_SOUNDFX_H +#define SOUND_MODS_SOUNDFX_H + +#include "common/stream.h" + +namespace Audio { + +class AudioStream; + +AudioStream *makeSoundFxStream(Common::SeekableReadStream *data, int rate = 44100, bool stereo = true); + +} // End of namespace Audio + +#endif Property changes on: scummvm/trunk/sound/mods/soundfx.h ___________________________________________________________________ Name: svn:mime-type + text/plain Name: svn:keywords + Date Rev Author URL Id Name: svn:eol-style + native Modified: scummvm/trunk/sound/module.mk =================================================================== --- scummvm/trunk/sound/module.mk 2007-05-18 21:12:23 UTC (rev 26868) +++ scummvm/trunk/sound/module.mk 2007-05-19 12:06:51 UTC (rev 26869) @@ -25,6 +25,7 @@ mods/protracker.o \ mods/paula.o \ mods/rjp1.o \ + mods/soundfx.o \ softsynth/adlib.o \ softsynth/ym2612.o \ softsynth/fluidsynth.o \ This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.

[Scummvm-cvs-logs] SF.net SVN: scummvm: [27468] scummvm/trunk/sound From: <sev@us...> - 2007-06-16 16:52 Revision: 27468 http://scummvm.svn.sourceforge.net/scummvm/?rev=27468&view=rev Author: sev Date: 2007-06-16 09:52:52 -0700 (Sat, 16 Jun 2007) Log Message: ----------- Fix legal header for new files. Modified Paths: -------------- scummvm/trunk/sound/rate_arm.cpp scummvm/trunk/sound/rate_arm_asm.s Modified: scummvm/trunk/sound/rate_arm.cpp =================================================================== --- scummvm/trunk/sound/rate_arm.cpp 2007-06-16 16:46:13 UTC (rev 27467) +++ scummvm/trunk/sound/rate_arm.cpp 2007-06-16 16:52:52 UTC (rev 27468) @@ -1,6 +1,9 @@ -/* ScummVM - Scumm Interpreter - * Copyright (C) 2001-2006 The ScummVM project +/* 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 Modified: scummvm/trunk/sound/rate_arm_asm.s =================================================================== --- scummvm/trunk/sound/rate_arm_asm.s 2007-06-16 16:46:13 UTC (rev 27467) +++ scummvm/trunk/sound/rate_arm_asm.s 2007-06-16 16:52:52 UTC (rev 27468) @@ -1,6 +1,9 @@ -@ ScummVM Scumm Interpreter -@ Copyright (C) 2007 The ScummVM project +@ 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 @@ -15,8 +18,8 @@ @ 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: $ +@ $URL$ +@ $Id$ @ @ @author Robin Watts (robin@...) @ This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.

[Scummvm-cvs-logs] SF.net SVN: scummvm: [27500] scummvm/trunk/sound From: <fingolfin@us...> - 2007-06-16 23:41 Revision: 27500 http://scummvm.svn.sourceforge.net/scummvm/?rev=27500&view=rev Author: fingolfin Date: 2007-06-16 16:41:37 -0700 (Sat, 16 Jun 2007) Log Message: ----------- cleanup Modified Paths: -------------- scummvm/trunk/sound/rate.cpp scummvm/trunk/sound/rate.h Modified: scummvm/trunk/sound/rate.cpp =================================================================== --- scummvm/trunk/sound/rate.cpp 2007-06-16 23:07:25 UTC (rev 27499) +++ scummvm/trunk/sound/rate.cpp 2007-06-16 23:41:37 UTC (rev 27500) @@ -78,7 +78,7 @@ SimpleRateConverter(st_rate_t inrate, st_rate_t outrate); int flow(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r); int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) { - return (ST_SUCCESS); + return ST_SUCCESS; } }; @@ -128,7 +128,7 @@ inPtr = inBuf; inLen = input.readBuffer(inBuf, ARRAYSIZE(inBuf)); if (inLen <= 0) - goto the_end; + return ST_EOF; } inLen -= (stereo ? 2 : 1); opos--; @@ -152,8 +152,7 @@ obuf += 2; } -the_end: - return (ST_SUCCESS); + return ST_SUCCESS; } /** @@ -189,7 +188,7 @@ LinearRateConverter(st_rate_t inrate, st_rate_t outrate); int flow(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r); int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) { - return (ST_SUCCESS); + return ST_SUCCESS; } }; @@ -244,7 +243,7 @@ inPtr = inBuf; inLen = input.readBuffer(inBuf, ARRAYSIZE(inBuf)); if (inLen <= 0) - goto the_end; + return ST_EOF; } inLen -= (stereo ? 2 : 1); ilast0 = icur0; @@ -258,10 +257,10 @@ // Loop as long as the outpos trails behind, and as long as there is // still space in the output buffer. - while (0 > opos) { + while (0 > opos && obuf < oend) { // interpolate st_sample_t out0, out1; - out0 = (st_sample_t)(ilast0 + (((icur0 - ilast0) * opos_frac + (1UL << (FRAC_BITS-1))) >> FRAC_BITS)); + out0 = (st_sample_t)(ilast0 + (((icur0 - ilast0) * opos_frac + (1UL << (FRAC_BITS-1))) >> FRAC_BITS)); out1 = (stereo ? (st_sample_t)(ilast1 + (((icur1 - ilast1) * opos_frac + (1UL << (FRAC_BITS-1))) >> FRAC_BITS)) : out0); @@ -278,14 +277,9 @@ long tmp = opos_frac + opos_inc_frac; opos += opos_inc + (tmp >> FRAC_BITS); opos_frac = tmp & ((1UL << FRAC_BITS) - 1); - - // Abort if we reached the end of the output buffer - if (obuf >= oend) - goto the_end; } } -the_end: - return (ST_SUCCESS); + return ST_SUCCESS; } @@ -339,11 +333,11 @@ obuf += 2; } - return (ST_SUCCESS); + return ST_SUCCESS; } virtual int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) { - return (ST_SUCCESS); + return ST_SUCCESS; } }; Modified: scummvm/trunk/sound/rate.h =================================================================== --- scummvm/trunk/sound/rate.h 2007-06-16 23:07:25 UTC (rev 27499) +++ scummvm/trunk/sound/rate.h 2007-06-16 23:41:37 UTC (rev 27500) @@ -40,11 +40,15 @@ typedef uint32 st_rate_t; /* Minimum and maximum values a sample can hold. */ -#define ST_SAMPLE_MAX 0x7fffL -#define ST_SAMPLE_MIN (-ST_SAMPLE_MAX - 1L) +enum { + ST_SAMPLE_MAX = 0x7fffL, + ST_SAMPLE_MIN = (-ST_SAMPLE_MAX - 1L) +}; -#define ST_EOF (-1) -#define ST_SUCCESS (0) +enum { + ST_EOF = -1, + ST_SUCCESS = 0 +}; static inline void clampedAdd(int16& a, int b) { register int val; This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.

[Scummvm-cvs-logs] SF.net SVN: scummvm: [27789] scummvm/trunk/sound From: <robinwatts@us...> - 2007-06-30 12:54 Revision: 27789 http://scummvm.svn.sourceforge.net/scummvm/?rev=27789&view=rev Author: robinwatts Date: 2007-06-30 05:54:07 -0700 (Sat, 30 Jun 2007) Log Message: ----------- Update the ARM version of the rate conversion code, in line with Fingolfins optimisatins of the C version last night. Saves about 5 cycles per sample. Modified Paths: -------------- scummvm/trunk/sound/rate_arm.cpp scummvm/trunk/sound/rate_arm_asm.s Modified: scummvm/trunk/sound/rate_arm.cpp =================================================================== --- scummvm/trunk/sound/rate_arm.cpp 2007-06-30 12:43:53 UTC (rev 27788) +++ scummvm/trunk/sound/rate_arm.cpp 2007-06-30 12:54:07 UTC (rev 27789) @@ -50,13 +50,18 @@ #include "sound/mixer.h" #include "common/util.h" +//#define DEBUG_RATECONV + namespace Audio { /** * The precision of the fractional computations used by the rate converter. * Normally you should never have to modify this value. + * This stuff is defined in common/frac.h, but we redefine it here as the + * ARM routine we call doesn't respect those definitions. */ #define FRAC_BITS 16 +#define FRAC_ONE (1<<FRAC_BITS) /** * The size of the intermediate input cache. Bigger values may increase @@ -152,6 +157,10 @@ extern "C" int SimpleRate_readFudge(Audio::AudioStream &input, int16 *a, int b) { +#ifdef DEBUG_RATECONV + fprintf(stderr, "Reading ptr=%x n%d\n", a, b); + fflush(stderr); +#endif return input.readBuffer(a, b); } @@ -197,7 +206,7 @@ int inLen; /** position of how far output is ahead of input */ - /** Holds what would have been opos-ipos */ + /** Holds what would have been opos-ipos<<16 + opos_frac */ long opos; /** integer position increment in the output stream */ @@ -206,14 +215,9 @@ /** current sample(s) in the input stream (left/right channel) */ st_sample_t icur[2]; /** last sample(s) in the input stream (left/right channel) */ - st_sample_t ilast[2]; + /** Note, these are deliberately ints, not st_sample_t's */ + int32 ilast[2]; - /** fractional position in the output stream */ - long opos_frac; - - /** fractional position increment in the output stream */ - long opos_inc_frac; - st_sample_t inBuf[INTERMEDIATE_BUFFER_SIZE]; } LinearRateDetails; @@ -270,16 +274,14 @@ error("rate effect can only handle rates < 65536"); } - lr.opos_frac = 0; - lr.opos = 1; + lr.opos = FRAC_ONE; /* increment */ incr = (inrate << FRAC_BITS) / outrate; - lr.opos_inc_frac = incr & ((1UL << FRAC_BITS) - 1); - lr.opos_inc = incr >> FRAC_BITS; + lr.opos_inc = incr; - lr.ilast[0] = lr.ilast[1] = 0; + lr.ilast[0] = lr.ilast[1] = 32768; lr.icur[0] = lr.icur[1] = 0; lr.inLen = 0; Modified: scummvm/trunk/sound/rate_arm_asm.s =================================================================== --- scummvm/trunk/sound/rate_arm_asm.s 2007-06-30 12:43:53 UTC (rev 27788) +++ scummvm/trunk/sound/rate_arm_asm.s 2007-06-30 12:54:07 UTC (rev 27789) @@ -396,6 +396,7 @@ LDMIA r2,{r0,r1,r8} @ r0 = inPtr @ r1 = inLen @ r8 = opos + MOV r10,#0 CMP r11,#0 @ if (osamp <= 0) BLE LinearRate_M_end @ bale ORR r12,r12,r12,LSL #8 @ r12= vol_l as 16 bits @@ -408,29 +409,23 @@ SUBS r1, r1, #1 @ r1 = inLen -= 1 BLT LinearRate_M_read LinearRate_M_read_return: - LDR r10,[r2, #16] @ r10= icur[0,1] + LDRH r4, [r2, #16] @ r4 = icur[0] LDRSH r5, [r0],#2 @ r5 = tmp1 = *inPtr++ - SUBS r8, r8, #1 @ r8 = opos-- - STR r10,[r2,#20] @ ilast[0,1] = icur[0,1] + SUBS r8, r8, #65536 @ r8 = opos-- + STRH r4, [r2,#22] @ ilast[0] = icur[0] STRH r5, [r2,#16] @ icur[0] = tmp1 BGE LinearRate_M_loop @ part2 - form output samples LinearRate_M_part2: @ We are guaranteed that opos < 0 here - LDRSH r6, [r2,#20] @ r6 = ilast[0] + LDR r6, [r2,#20] @ r6 = ilast[0]<<16 + 32768 LDRSH r5, [r2,#16] @ r5 = icur[0] - LDRH r4, [r2,#24] @ r4 = opos_frac - LDR r10,[r2,#28] @ r10= opos_frac_inc - MOV r6, r6, LSL #16 @ r6 = ilast[0]<<16 + MOV r4, r8, LSL #16 + MOV r4, r4, LSR #16 SUB r5, r5, r6, ASR #16 @ r5 = icur[0] - ilast[0] - ADD r6, r6, #1<<15 @ r6 = ilast[0]+1<<(FRAC_BITS-1) MLA r6, r4, r5, r6 @ r6 = (icur[0]-ilast[0])*opos_frac+ilast[0] - ADD r4, r4, r10 @ r4 = tmp = opos_frac+opos_inc_frac - STRH r4,[r2,#24] @ opos_frac &= 65535 - ADD r8, r8, r4, LSR #16 @ opos += (tmp>>FRAC_BITS) - LDRSH r4, [r3] @ r4 = obuf[0] LDRSH r5, [r3,#2] @ r5 = obuf[1] MOV r6, r6, ASR #16 @ r6 = tmp0 = tmp1 >>= 16 @@ -438,10 +433,9 @@ MUL r6, r14,r6 @ r6 = tmp1*vol_r ADDS r7, r7, r4, LSL #16 @ r7 = obuf[0]<<16 + tmp0*vol_l - MOV r4, #0 - RSCVS r7, r4, #1<<31 @ Clamp r7 + RSCVS r7, r10, #1<<31 @ Clamp r7 ADDS r6, r6, r5, LSL #16 @ r6 = obuf[1]<<16 + tmp1*vol_r - RSCVS r6, r4, #1<<31 @ Clamp r6 + RSCVS r6, r10, #1<<31 @ Clamp r6 MOV r7, r7, LSR #16 @ Shift back to halfword MOV r6, r6, LSR #16 @ Shift back to halfword @@ -449,7 +443,7 @@ LDR r5, [r2,#12] @ r5 = opos_inc STRH r7, [r3],#2 @ Store output value STRH r6, [r3],#2 @ Store output value - SUBS r11, r11,#1 @ opos-- + SUBS r11, r11,#1 @ osamp-- BLE LinearRate_M_end @ end if needed ADDS r8, r8, r5 @ r8 = opos += opos_inc @@ -460,7 +454,7 @@ STMIA r2,{r0,r1,r8} LDMFD r13!,{r4-r11,PC} LinearRate_M_read: - ADD r0, r2, #32 @ r0 = inPtr = inBuf + ADD r0, r2, #28 @ r0 = inPtr = inBuf STMFD r13!,{r0,r2-r3,r12,r14} MOV r1, r0 @ r1 = inBuf @@ -508,43 +502,37 @@ LDR r10,[r2, #16] @ r10= icur[0,1] LDRSH r5, [r0],#2 @ r5 = tmp0 = *inPtr++ LDRSH r6, [r0],#2 @ r5 = tmp1 = *inPtr++ - SUBS r8, r8, #1 @ r8 = opos-- - STR r10,[r2,#20] @ ilast[0,1] = icur[0,1] + SUBS r8, r8, #65536 @ r8 = opos-- + STRH r10,[r2,#22] @ ilast[0] = icur[0] + MOV r10,r10,LSR #16 + STRH r10,[r2,#26] @ ilast[1] = icur[1] STRH r5, [r2,#16] @ icur[0] = tmp0 - STRH r6, [r2,#16] @ icur[1] = tmp1 + STRH r6, [r2,#18] @ icur[1] = tmp1 BGE LinearRate_S_loop @ part2 - form output samples LinearRate_S_part2: @ We are guaranteed that opos < 0 here - LDRSH r6, [r2,#20] @ r6 = ilast[0] + LDR r6, [r2,#20] @ r6 = ilast[0]<<16 + 32768 LDRSH r5, [r2,#16] @ r5 = icur[0] - LDRH r4, [r2,#24] @ r4 = opos_frac - MOV r6, r6, LSL #16 @ r6 = ilast[0]<<16 + MOV r4, r8, LSL #16 + MOV r4, r4, LSR #16 SUB r5, r5, r6, ASR #16 @ r5 = icur[0] - ilast[0] - ADD r6, r6, #1<<15 @ r6 = ilast[0]+1<<(FRAC_BITS-1) MLA r6, r4, r5, r6 @ r6 = (icur[0]-ilast[0])*opos_frac+ilast[0] - LDRSH r7, [r2,#22] @ r6 = ilast[1] + LDR r7, [r2,#24] @ r7 = ilast[1]<<16 + 32768 LDRSH r5, [r2,#18] @ r5 = icur[1] - LDR r10,[r2,#28] @ r10= opos_frac_inc - MOV r7, r7, LSL #16 @ r7 = ilast[1]<<16 + LDRSH r10,[r3] @ r10= obuf[0] + MOV r6, r6, ASR #16 @ r6 = tmp1 >>= 16 SUB r5, r5, r7, ASR #16 @ r5 = icur[1] - ilast[1] - ADD r7, r7, #1<<15 @ r6 = ilast[1]+1<<(FRAC_BITS-1) - MLA r7, r4, r5, r7 @ r6 = (icur[1]-ilast[1])*opos_frac+ilast[1] + MLA r7, r4, r5, r7 @ r7 = (icur[1]-ilast[1])*opos_frac+ilast[1] - ADD r4, r4, r10 @ r4 = tmp = opos_frac+opos_inc_frac - STRH r4,[r2,#24] @ opos_frac &= 65535 - ADD r8, r8, r4, LSR #16 @ opos += (tmp>>FRAC_BITS) - - LDRSH r4, [r3] @ r4 = obuf[0] LDRSH r5, [r3,#2] @ r5 = obuf[1] MOV r7, r7, ASR #16 @ r7 = tmp0 >>= 16 - MOV r6, r6, ASR #16 @ r6 = tmp1 >>= 16 MUL r7, r12,r7 @ r7 = tmp0*vol_l MUL r6, r14,r6 @ r6 = tmp1*vol_r - ADDS r7, r7, r4, LSL #16 @ r7 = obuf[0]<<16 + tmp0*vol_l + ADDS r7, r7, r10, LSL #16 @ r7 = obuf[0]<<16 + tmp0*vol_l MOV r4, #0 RSCVS r7, r4, #1<<31 @ Clamp r7 ADDS r6, r6, r5, LSL #16 @ r6 = obuf[1]<<16 + tmp1*vol_r @@ -556,7 +544,7 @@ LDR r5, [r2,#12] @ r5 = opos_inc STRH r7, [r3],#2 @ Store output value STRH r6, [r3],#2 @ Store output value - SUBS r11, r11,#1 @ opos-- + SUBS r11, r11,#1 @ osamp-- BLE LinearRate_S_end @ and loop ADDS r8, r8, r5 @ r8 = opos += opos_inc @@ -567,7 +555,7 @@ STMIA r2,{r0,r1,r8} LDMFD r13!,{r4-r11,PC} LinearRate_S_read: - ADD r0, r2, #32 @ r0 = inPtr = inBuf + ADD r0, r2, #28 @ r0 = inPtr = inBuf STMFD r13!,{r0,r2-r3,r12,r14} MOV r1, r0 @ r1 = inBuf @@ -615,43 +603,37 @@ LDR r10,[r2, #16] @ r10= icur[0,1] LDRSH r5, [r0],#2 @ r5 = tmp0 = *inPtr++ LDRSH r6, [r0],#2 @ r5 = tmp1 = *inPtr++ - SUBS r8, r8, #1 @ r8 = opos-- - STR r10,[r2,#20] @ ilast[0,1] = icur[0,1] + SUBS r8, r8, #65536 @ r8 = opos-- + STRH r10,[r2,#22] @ ilast[0] = icur[0] + MOV r10,r10,LSR #16 + STRH r10,[r2,#22] @ ilast[1] = icur[1] STRH r5, [r2,#16] @ icur[0] = tmp0 - STRH r6, [r2,#16] @ icur[1] = tmp1 + STRH r6, [r2,#18] @ icur[1] = tmp1 BGE LinearRate_R_loop @ part2 - form output samples LinearRate_R_part2: @ We are guaranteed that opos < 0 here - LDRSH r6, [r2,#20] @ r6 = ilast[0] + LDR r6, [r2,#20] @ r6 = ilast[0] LDRSH r5, [r2,#16] @ r5 = icur[0] - LDRH r4, [r2,#24] @ r4 = opos_frac - MOV r6, r6, LSL #16 @ r6 = ilast[0]<<16 + MOV r4, r8, LSL #16 + MOV r4, r4, LSR #16 SUB r5, r5, r6, ASR #16 @ r5 = icur[0] - ilast[0] - ADD r6, r6, #1<<15 @ r6 = ilast[0]+1<<(FRAC_BITS-1) MLA r6, r4, r5, r6 @ r6 = (icur[0]-ilast[0])*opos_frac+ilast[0] - LDRSH r7, [r2,#22] @ r6 = ilast[1] + LDR r7, [r2,#24] @ r7 = ilast[1] LDRSH r5, [r2,#18] @ r5 = icur[1] - LDR r10,[r2,#28] @ r10= opos_frac_inc - MOV r7, r7, LSL #16 @ r7 = ilast[1]<<16 + LDR r10,[r3] @ r10= obuf[0] + MOV r6, r6, ASR #16 @ r6 = tmp1 >>= 16 SUB r5, r5, r7, ASR #16 @ r5 = icur[1] - ilast[1] - ADD r7, r7, #1<<15 @ r6 = ilast[1]+1<<(FRAC_BITS-1) - MLA r7, r4, r5, r7 @ r6 = (icur[1]-ilast[1])*opos_frac+ilast[1] + MLA r7, r4, r5, r7 @ r7 = (icur[1]-ilast[1])*opos_frac+ilast[1] - ADD r4, r4, r10 @ r4 = tmp = opos_frac+opos_inc_frac - STRH r4,[r2,#24] @ opos_frac &= 65535 - ADD r8, r8, r4, LSR #16 @ opos += (tmp>>FRAC_BITS) - - LDRSH r4, [r3] @ r4 = obuf[0] LDRSH r5, [r3,#2] @ r5 = obuf[1] MOV r7, r7, ASR #16 @ r7 = tmp0 >>= 16 - MOV r6, r6, ASR #16 @ r6 = tmp1 >>= 16 MUL r7, r12,r7 @ r7 = tmp0*vol_l MUL r6, r14,r6 @ r6 = tmp1*vol_r - ADDS r7, r7, r4, LSL #16 @ r7 = obuf[0]<<16 + tmp0*vol_l + ADDS r7, r7, r10, LSL #16 @ r7 = obuf[0]<<16 + tmp0*vol_l MOV r4, #0 RSCVS r7, r4, #1<<31 @ Clamp r7 ADDS r6, r6, r5, LSL #16 @ r6 = obuf[1]<<16 + tmp1*vol_r @@ -663,7 +645,7 @@ LDR r5, [r2,#12] @ r5 = opos_inc STRH r6, [r3],#2 @ Store output value STRH r7, [r3],#2 @ Store output value - SUBS r11, r11,#1 @ opos-- + SUBS r11, r11,#1 @ osamp-- BLE LinearRate_R_end @ and loop ADDS r8, r8, r5 @ r8 = opos += opos_inc @@ -674,7 +656,7 @@ STMIA r2,{r0,r1,r8} LDMFD r13!,{r4-r11,PC} LinearRate_R_read: - ADD r0, r2, #32 @ r0 = inPtr = inBuf + ADD r0, r2, #28 @ r0 = inPtr = inBuf STMFD r13!,{r0,r2-r3,r12,r14} MOV r1, r0 @ r1 = inBuf This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.

[Scummvm-cvs-logs] SF.net SVN: scummvm: [29682] scummvm/trunk/sound From: <thebluegr@us...> - 2007-12-01 16:31 Revision: 29682 http://scummvm.svn.sourceforge.net/scummvm/?rev=29682&view=rev Author: thebluegr Date: 2007-12-01 08:31:10 -0800 (Sat, 01 Dec 2007) Log Message: ----------- Slight cleanup and commenting of the ADPCM decoder Modified Paths: -------------- scummvm/trunk/sound/adpcm.cpp scummvm/trunk/sound/adpcm.h Modified: scummvm/trunk/sound/adpcm.cpp =================================================================== --- scummvm/trunk/sound/adpcm.cpp 2007-12-01 16:18:26 UTC (rev 29681) +++ scummvm/trunk/sound/adpcm.cpp 2007-12-01 16:31:10 UTC (rev 29682) @@ -265,10 +265,10 @@ predictor = (((c->sample1) * (c->coeff1)) + ((c->sample2) * (c->coeff2))) / 256; predictor += (signed)((code & 0x08) ? (code - 0x10) : (code)) * c->delta; - if (predictor < -0x8000) - predictor = -0x8000; - else if (predictor > 0x7fff) - predictor = 0x7fff; + if (predictor < -32768) + predictor = -32768; + else if (predictor > 32767) + predictor = 32767; c->sample2 = c->sample1; c->sample1 = predictor; @@ -345,13 +345,12 @@ diff = (code & 0x08) ? -E : E; samp = _status.last + diff; - if (samp < -0x8000) - samp = -0x8000; - else if (samp > 0x7fff) - samp = 0x7fff; + if (samp < -32768) + samp = -32768; + else if (samp > 32767) + samp = 32767; _status.last = samp; - _status.stepIndex += stepAdjust(code); if (_status.stepIndex < 0) _status.stepIndex = 0; Modified: scummvm/trunk/sound/adpcm.h =================================================================== --- scummvm/trunk/sound/adpcm.h 2007-12-01 16:18:26 UTC (rev 29681) +++ scummvm/trunk/sound/adpcm.h 2007-12-01 16:31:10 UTC (rev 29682) @@ -34,10 +34,15 @@ class AudioStream; +// There are several types of ADPCM encoding, only some are supported here +// For all the different encodings, refer to: +// http://wiki.multimedia.cx/index.php?title=Category:ADPCM_Audio_Codecs +// Usually, if the audio stream we're trying to play has the FourCC header +// string intact, it's easy to discern which encoding is used enum typesADPCM { - kADPCMOki, - kADPCMMSIma, - kADPCMMS + kADPCMOki, // Dialogic/Oki ADPCM (aka VOX) + kADPCMMSIma, // Microsoft IMA ADPCM + kADPCMMS // Microsoft ADPCM }; /** This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.

[Scummvm-cvs-logs] SF.net SVN: scummvm: [32258] scummvm/trunk/sound From: <fingolfin@us...> - 2008-05-24 22:41 Revision: 32258 http://scummvm.svn.sourceforge.net/scummvm/?rev=32258&view=rev Author: fingolfin Date: 2008-05-24 15:41:15 -0700 (Sat, 24 May 2008) Log Message: ----------- Patch #1970427: AudioStream play length querying Modified Paths: -------------- scummvm/trunk/sound/audiostream.cpp scummvm/trunk/sound/audiostream.h Modified: scummvm/trunk/sound/audiostream.cpp =================================================================== --- scummvm/trunk/sound/audiostream.cpp 2008-05-24 22:11:41 UTC (rev 32257) +++ scummvm/trunk/sound/audiostream.cpp 2008-05-24 22:41:15 UTC (rev 32258) @@ -102,6 +102,11 @@ #pragma mark --- LinearMemoryStream --- #pragma mark - +inline int32 calculatePlayTime(int rate, int samples) { + int32 seconds = samples / rate; + int32 milliseconds = (1000 * (samples % rate)) / rate; + return seconds * 1000 + milliseconds; +} /** * A simple raw audio stream, purely memory based. It operates on a single @@ -122,10 +127,11 @@ const byte *_loopEnd; const int _rate; const byte *_origPtr; + const int32 _playtime; 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) { + : _ptr(ptr), _end(ptr+len), _loopPtr(0), _loopEnd(0), _rate(rate), _playtime(calculatePlayTime(rate, len / (is16Bit ? 2 : 1) / (stereo ? 2 : 1))) { // Verify the buffer sizes are sane if (is16Bit && stereo) @@ -147,10 +153,11 @@ } int readBuffer(int16 *buffer, const int numSamples); - bool isStereo() const { return stereo; } - bool endOfData() const { return _ptr >= _end; } + bool isStereo() const { return stereo; } + bool endOfData() const { return _ptr >= _end; } - int getRate() const { return _rate; } + int getRate() const { return _rate; } + int32 getTotalPlayTime() const { return _playtime; } }; template<bool stereo, bool is16Bit, bool isUnsigned, bool isLE> Modified: scummvm/trunk/sound/audiostream.h =================================================================== --- scummvm/trunk/sound/audiostream.h 2008-05-24 22:11:41 UTC (rev 32257) +++ scummvm/trunk/sound/audiostream.h 2008-05-24 22:41:15 UTC (rev 32258) @@ -93,6 +93,20 @@ * NULL in case of an error (e.g. invalid/nonexisting file) */ static AudioStream* openStreamFile(const Common::String &basename, uint32 startTime = 0, uint32 duration = 0, uint numLoops = 1); + + enum { + kUnknownPlayTime = -1 + }; + + /** + * Returns total playtime of the AudioStream object. + * Note that this does not require to return an playtime, if the + * playtime of the AudioStream is unknown it returns 'kUnknownPlayTime'. + * @see kUnknownPlayTime + * + * @return playtime in milliseconds + */ + int32 getTotalPlayTime() const { return kUnknownPlayTime; } }; /** This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.

[Scummvm-cvs-logs] SF.net SVN: scummvm: [32263] scummvm/trunk/sound From: <lordhoto@us...> - 2008-05-24 23:39 Revision: 32263 http://scummvm.svn.sourceforge.net/scummvm/?rev=32263&view=rev Author: lordhoto Date: 2008-05-24 16:39:13 -0700 (Sat, 24 May 2008) Log Message: ----------- Implementation of AudioStream::getTotalPlayTime for FLAC and OGG/Vorbis streams. Modified Paths: -------------- scummvm/trunk/sound/flac.cpp scummvm/trunk/sound/vorbis.cpp Modified: scummvm/trunk/sound/flac.cpp =================================================================== --- scummvm/trunk/sound/flac.cpp 2008-05-24 23:05:01 UTC (rev 32262) +++ scummvm/trunk/sound/flac.cpp 2008-05-24 23:39:13 UTC (rev 32263) @@ -141,6 +141,26 @@ return _streaminfo.channels == 0 || (_lastSampleWritten && _sampleCache.bufFill == 0); } + int32 getTotalPlayTime() const { + int32 samples = 0; + + if (!_lastSample) { + if (!_streaminfo.total_samples) + return AudioStream::kUnknownPlayTime; + + samples = _streaminfo.total_samples - _firstSample; + } else { + samples = _lastSample - _firstSample - 1; + } + + const int32 rate = _streaminfo.sample_rate; + + int32 seconds = samples / rate; + int32 milliseconds = (1000 * (samples % rate)) / rate; + + return seconds * 1000 + milliseconds; + } + bool isStreamDecoderReady() const { return getStreamDecoderState() == FLAC__STREAM_DECODER_SEARCH_FOR_FRAME_SYNC ; } protected: Modified: scummvm/trunk/sound/vorbis.cpp =================================================================== --- scummvm/trunk/sound/vorbis.cpp 2008-05-24 23:05:01 UTC (rev 32262) +++ scummvm/trunk/sound/vorbis.cpp 2008-05-24 23:39:13 UTC (rev 32263) @@ -118,6 +118,14 @@ bool isStereo() const { return _isStereo; } int getRate() const { return _rate; } + int32 getTotalPlayTime() const { +#ifdef USE_TREMOR + return _endTime - _startTime; +#else + return (int32)((_endTime - _startTime) * 1000.0); +#endif + } + protected: void refill(); }; This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.

[Scummvm-cvs-logs] SF.net SVN: scummvm: [32264] scummvm/trunk/sound From: <lordhoto@us...> - 2008-05-24 23:49 Revision: 32264 http://scummvm.svn.sourceforge.net/scummvm/?rev=32264&view=rev Author: lordhoto Date: 2008-05-24 16:49:15 -0700 (Sat, 24 May 2008) Log Message: ----------- Take numLoops in account for getTotalPlayTime in FLAC and OGG/Vorbis streams. Modified Paths: -------------- scummvm/trunk/sound/flac.cpp scummvm/trunk/sound/vorbis.cpp Modified: scummvm/trunk/sound/flac.cpp =================================================================== --- scummvm/trunk/sound/flac.cpp 2008-05-24 23:39:13 UTC (rev 32263) +++ scummvm/trunk/sound/flac.cpp 2008-05-24 23:49:15 UTC (rev 32264) @@ -90,6 +90,7 @@ bool _disposeAfterUse; uint _numLoops; + const uint _totalNumLoops; ::FLAC__SeekableStreamDecoder *_decoder; @@ -158,7 +159,7 @@ int32 seconds = samples / rate; int32 milliseconds = (1000 * (samples % rate)) / rate; - return seconds * 1000 + milliseconds; + return (seconds * 1000 + milliseconds) * _totalNumLoops; } bool isStreamDecoderReady() const { return getStreamDecoderState() == FLAC__STREAM_DECODER_SEARCH_FOR_FRAME_SYNC ; } @@ -210,6 +211,7 @@ _inStream(inStream), _disposeAfterUse(dispose), _numLoops(numLoops), + _totalNumLoops(numLoops), _firstSample(0), _lastSample(0), _outBuffer(NULL), _requestedSamples(0), _lastSampleWritten(false), _methodConvertBuffers(&FlacInputStream::convertBuffersGeneric) Modified: scummvm/trunk/sound/vorbis.cpp =================================================================== --- scummvm/trunk/sound/vorbis.cpp 2008-05-24 23:39:13 UTC (rev 32263) +++ scummvm/trunk/sound/vorbis.cpp 2008-05-24 23:49:15 UTC (rev 32264) @@ -92,6 +92,7 @@ bool _isStereo; int _rate; uint _numLoops; + const uint _totalNumLoops; #ifdef USE_TREMOR ogg_int64_t _startTime; @@ -120,9 +121,9 @@ int32 getTotalPlayTime() const { #ifdef USE_TREMOR - return _endTime - _startTime; + return (_endTime - _startTime) * _totalNumLoops; #else - return (int32)((_endTime - _startTime) * 1000.0); + return (int32)((_endTime - _startTime) * 1000.0) * _totalNumLoops; #endif } @@ -134,6 +135,7 @@ _inStream(inStream), _disposeAfterUse(dispose), _numLoops(numLoops), + _totalNumLoops(numLoops), _bufferEnd(_buffer + ARRAYSIZE(_buffer)) { bool err = (ov_open_callbacks(inStream, &_ovFile, NULL, 0, g_stream_wrap) < 0); This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.

[Scummvm-cvs-logs] SF.net SVN: scummvm: [32265] scummvm/trunk/sound From: <lordhoto@us...> - 2008-05-24 23:58 Revision: 32265 http://scummvm.svn.sourceforge.net/scummvm/?rev=32265&view=rev Author: lordhoto Date: 2008-05-24 16:58:22 -0700 (Sat, 24 May 2008) Log Message: ----------- Instead of simply returning 0 for "infinite" looping in FLAC and OGG/Vorbis streams in getTotalPlayTime return kUnknownPlayTime. Modified Paths: -------------- scummvm/trunk/sound/flac.cpp scummvm/trunk/sound/vorbis.cpp Modified: scummvm/trunk/sound/flac.cpp =================================================================== --- scummvm/trunk/sound/flac.cpp 2008-05-24 23:49:15 UTC (rev 32264) +++ scummvm/trunk/sound/flac.cpp 2008-05-24 23:58:22 UTC (rev 32265) @@ -143,6 +143,9 @@ } int32 getTotalPlayTime() const { + if (!_totalNumLoops) + return AudioStream::kUnknownPlayTime; + int32 samples = 0; if (!_lastSample) { Modified: scummvm/trunk/sound/vorbis.cpp =================================================================== --- scummvm/trunk/sound/vorbis.cpp 2008-05-24 23:49:15 UTC (rev 32264) +++ scummvm/trunk/sound/vorbis.cpp 2008-05-24 23:58:22 UTC (rev 32265) @@ -120,6 +120,9 @@ int getRate() const { return _rate; } int32 getTotalPlayTime() const { + if (!_totalNumLoops) + return AudioStream::kUnknownPlayTime; + #ifdef USE_TREMOR return (_endTime - _startTime) * _totalNumLoops; #else This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.

[Scummvm-cvs-logs] SF.net SVN: scummvm: [32323] scummvm/trunk/sound From: <lordhoto@us...> - 2008-05-27 13:28 Revision: 32323 http://scummvm.svn.sourceforge.net/scummvm/?rev=32323&view=rev Author: lordhoto Date: 2008-05-27 06:28:47 -0700 (Tue, 27 May 2008) Log Message: ----------- - Cleanup code for FlacInputStream::getTotalPlayTime - Reset input stream position in MP3InputStream after calculating total play time Modified Paths: -------------- scummvm/trunk/sound/flac.cpp scummvm/trunk/sound/mp3.cpp Modified: scummvm/trunk/sound/flac.cpp =================================================================== --- scummvm/trunk/sound/flac.cpp 2008-05-27 13:15:51 UTC (rev 32322) +++ scummvm/trunk/sound/flac.cpp 2008-05-27 13:28:47 UTC (rev 32323) @@ -90,7 +90,6 @@ bool _disposeAfterUse; uint _numLoops; - const uint _totalNumLoops; ::FLAC__SeekableStreamDecoder *_decoder; @@ -103,6 +102,9 @@ /** index + 1(!) of the last sample to be played - 0 is end of stream */ FLAC__uint64 _lastSample; + /** total play time */ + int32 _totalPlayTime; + /** true if the last sample was decoded from the FLAC-API - there might still be data in the buffer */ bool _lastSampleWritten; @@ -142,29 +144,8 @@ return _streaminfo.channels == 0 || (_lastSampleWritten && _sampleCache.bufFill == 0); } - int32 getTotalPlayTime() const { - if (!_totalNumLoops) - return AudioStream::kUnknownPlayTime; + int32 getTotalPlayTime() const { return _totalPlayTime; } - int32 samples = 0; - - if (!_lastSample) { - if (!_streaminfo.total_samples) - return AudioStream::kUnknownPlayTime; - - samples = _streaminfo.total_samples - _firstSample; - } else { - samples = _lastSample - _firstSample - 1; - } - - const int32 rate = _streaminfo.sample_rate; - - int32 seconds = samples / rate; - int32 milliseconds = (1000 * (samples % rate)) / rate; - - return (seconds * 1000 + milliseconds) * _totalNumLoops; - } - bool isStreamDecoderReady() const { return getStreamDecoderState() == FLAC__STREAM_DECODER_SEARCH_FOR_FRAME_SYNC ; } protected: @@ -214,7 +195,6 @@ _inStream(inStream), _disposeAfterUse(dispose), _numLoops(numLoops), - _totalNumLoops(numLoops), _firstSample(0), _lastSample(0), _outBuffer(NULL), _requestedSamples(0), _lastSampleWritten(false), _methodConvertBuffers(&FlacInputStream::convertBuffersGeneric) @@ -260,7 +240,28 @@ // avoid overflows). _firstSample = (FLAC__uint64)(startTime * (_streaminfo.sample_rate / 1000.0)); _lastSample = (FLAC__uint64)(endTime * (_streaminfo.sample_rate / 1000.0)); + if (_firstSample == 0 || seekAbsolute(_firstSample)) { + int32 samples = kUnknownPlayTime; + + if (!_lastSample) { + if (_streaminfo.total_samples) + samples = _streaminfo.total_samples - _firstSample; + } else { + samples = _lastSample - _firstSample - 1; + } + + if (samples != kUnknownPlayTime && samples >= 0 && numLoops) { + const int32 rate = _streaminfo.sample_rate; + + int32 seconds = samples / rate; + int32 milliseconds = (1000 * (samples % rate)) / rate; + + _totalPlayTime = (seconds * 1000 + milliseconds) * numLoops; + } else { + _totalPlayTime = kUnknownPlayTime; + } + return; // no error occured } } Modified: scummvm/trunk/sound/mp3.cpp =================================================================== --- scummvm/trunk/sound/mp3.cpp 2008-05-27 13:15:51 UTC (rev 32322) +++ scummvm/trunk/sound/mp3.cpp 2008-05-27 13:28:47 UTC (rev 32323) @@ -169,11 +169,14 @@ // Reinit stream _state = MP3_STATE_INIT; + + // Reset the stream data + _inStream->seek(0, SEEK_SET); } _totalPlayTime = mad_timer_count(length, MAD_UNITS_MILLISECONDS); - if (numLoops) + if (numLoops && mad_timer_sign(length) >= 0) _totalPlayTime *= numLoops; else _totalPlayTime = kUnknownPlayTime; This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.

[Scummvm-cvs-logs] SF.net SVN: scummvm: [32973] scummvm/trunk/sound From: <fingolfin@us...> - 2008-07-09 10:42 Revision: 32973 http://scummvm.svn.sourceforge.net/scummvm/?rev=32973&view=rev Author: fingolfin Date: 2008-07-09 03:42:47 -0700 (Wed, 09 Jul 2008) Log Message: ----------- cleanup / code formatting Modified Paths: -------------- scummvm/trunk/sound/midiparser_smf.cpp scummvm/trunk/sound/midiparser_xmidi.cpp Modified: scummvm/trunk/sound/midiparser_smf.cpp =================================================================== --- scummvm/trunk/sound/midiparser_smf.cpp 2008-07-09 02:49:20 UTC (rev 32972) +++ scummvm/trunk/sound/midiparser_smf.cpp 2008-07-09 10:42:47 UTC (rev 32973) @@ -85,16 +85,26 @@ _position._running_status = info.event; switch (info.command()) { - case 0xC: case 0xD: + case 0x9: // Note On info.basic.param1 = *(_position._play_pos++); + info.basic.param2 = *(_position._play_pos++); + if (info.basic.param2 == 0) + info.event = info.channel() | 0x80; + info.length = 0; + break; + + case 0xC: + case 0xD: + info.basic.param1 = *(_position._play_pos++); info.basic.param2 = 0; break; - case 0x8: case 0x9: case 0xA: case 0xB: case 0xE: + case 0x8: + case 0xA: + case 0xB: + case 0xE: info.basic.param1 = *(_position._play_pos++); info.basic.param2 = *(_position._play_pos++); - if (info.command() == 0x9 && info.basic.param2 == 0) - info.event = info.channel() | 0x80; info.length = 0; break; @@ -110,7 +120,12 @@ info.basic.param2 = 0; break; - case 0x6: case 0x8: case 0xA: case 0xB: case 0xC: case 0xE: + case 0x6: + case 0x8: + case 0xA: + case 0xB: + case 0xC: + case 0xE: info.basic.param1 = info.basic.param2 = 0; break; Modified: scummvm/trunk/sound/midiparser_xmidi.cpp =================================================================== --- scummvm/trunk/sound/midiparser_xmidi.cpp 2008-07-09 02:49:20 UTC (rev 32972) +++ scummvm/trunk/sound/midiparser_xmidi.cpp 2008-07-09 10:42:47 UTC (rev 32973) @@ -86,11 +86,19 @@ } break; - case 0xC: case 0xD: + case 0xC: + case 0xD: info.basic.param1 = *(_position._play_pos++); info.basic.param2 = 0; break; + case 0x8: + case 0xA: + case 0xE: + info.basic.param1 = *(_position._play_pos++); + info.basic.param2 = *(_position._play_pos++); + break; + case 0xB: info.basic.param1 = *(_position._play_pos++); info.basic.param2 = *(_position._play_pos++); @@ -128,11 +136,6 @@ } break; - case 0x8: case 0xA: case 0xE: - info.basic.param1 = *(_position._play_pos++); - info.basic.param2 = *(_position._play_pos++); - break; - case 0xF: // Meta or SysEx event switch (info.event & 0x0F) { case 0x2: // Song Position Pointer @@ -145,7 +148,12 @@ info.basic.param2 = 0; break; - case 0x6: case 0x8: case 0xA: case 0xB: case 0xC: case 0xE: + case 0x6: + case 0x8: + case 0xA: + case 0xB: + case 0xC: + case 0xE: info.basic.param1 = info.basic.param2 = 0; break; This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.

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