Merge remote branch 'upstream/master' into pegasus

Conflicts:
	AUTHORS
	devtools/credits.pl
	gui/credits.h

AUTHORS

@@ -89,7 +89,9 @@ ScummVM Team
     DreamWeb:
        Torbjorn Andersson
        Bertrand Augereau
+       Filippos Karapetis
        Vladimir Menshakov    - (retired)
+       Willem Jan Palenstijn
 
     Gob:
        Torbjorn Andersson
@@ -421,10 +423,10 @@ Other contributions
        Matteo Angelino
 
     Norwegian (Bokmaal):
-       Einar Johan T. Somaae
+       Einar Johan Somaaen
 
     Norwegian (Nynorsk):
-       Einar Johan T. Somaae
+       Einar Johan Somaaen
 
     Polish:
        GrajPoPolsku.pl Team
@@ -600,6 +602,13 @@ Special thanks to
    Broken Sword 2.5 team for providing sources of their engine and their
    great support.
 
+   Neil Dodwell and David Dew from Creative Reality for providing the source
+   of Dreamweb and for their tremendous support.
+
+   Janusz Wisniewski and Miroslaw Liminowicz from Laboratorium Komputerowe
+   Avalon for providing full source code for Soltys and letting us to
+   redistribute the game.
+
    Bob Bell, Michel Kripalani, Tommy Yune, from Presto Studios for providing
    the source code of The Journeyman Project: Pegasus Prime.
 

NEWS

@@ -1,18 +1,23 @@
 For a more comprehensive changelog of the latest experimental code, see:
         https://github.com/scummvm/scummvm/commits/
 
-1.5.0 (????-??-??)
+1.6.0 (????-??-??)
+ General:
+   - Added a new save/load chooser based on a grid of thumbnails. This is only
+     supported for resolutions bigger than 640x400. The old chooser is still
+     available and used for games without thumbnail support. It is possible to
+     select the old one as default too.
+   - Rewrote VideoDecoder subsystem.
+
+1.5.0 (2012-07-27)
  New Games:
    - Added support for Backyard Baseball 2003.
    - Added support for Blue Force.
    - Added support for Darby the Dragon.
    - Added support for Dreamweb.
    - Added support for Geisha.
    - Added support for Gregory and the Hot Air Balloon.
-   - Added support for Magic Tales: Baba Yaga and the Magic Geese.
-   - Added support for Magic Tales: Imo and the King.
    - Added support for Magic Tales: Liam Finds a Story.
-   - Added support for Magic Tales: The Little Samurai.
    - Added support for Once Upon A Time: Little Red Riding Hood
    - Added support for Sleeping Cub's Test of Courage.
    - Added support for Soltys.
@@ -33,19 +38,21 @@ For a more comprehensive changelog of the latest experimental code, see:
    - Improved predictive dialog look.
    - Various GUI improvements.
 
- SDL ports:
-   - Added support for OpenGL (GSoC Task).
-
  Broken Sword 1:
    - Fixed incorrect sound effects in the DOS/Windows demo.
    - Added support for PlayStation videos.
+   - Fixed missing subtitles in the demo.
 
  Broken Sword 2:
    - Added support for PlayStation videos.
 
  Cine:
    - Implemented Roland MT-32 output driver.
 
+ Drascula:
+   - Added Spanish subtitles in the Von Braun cutscene (#3069981: no subtitles
+     in scene with "von Braun").
+
  Gob:
    - Fixed a crash in Lost in Time
    - Rewrote the AdLib player. Enabled the now working MDY player in
@@ -63,6 +70,10 @@ For a more comprehensive changelog of the latest experimental code, see:
    - Added aspect ratio correction feature.
    - Implemented 16 bits per pixel support for games.
 
+ Maemo port:
+   - Added support for Nokia 770 running OS2008 HE.
+   - Added configurable keymap.
+
  Windows port:
    - Changed default savegames location for Windows NT4/2000/XP/Vista/7.
      (The migration batch file can be used to copy savegames from the old

README

@@ -2084,7 +2084,7 @@ Sierra games using the SCI engine add the following non-standard keywords:
     originalsaveload   bool     If true, the original save/load screens are
                                 used instead of the enhanced ScummVM ones
     native_fb01        bool     If true, the music driver for an IBM Music
-                                Feature card or a Yahama FB-01 FM synth module
+                                Feature card or a Yamaha FB-01 FM synth module
                                 is used for MIDI output
 
 Broken Sword II adds the following non-standard keywords:
@@ -2125,7 +2125,7 @@ Lands of Lore: The Throne of Chaos adds the following non-standard keywords:
 Space Quest IV CD adds the following non-standard keyword:
 
      silver_cursors    bool     If true, an alternate set of silver mouse cursors
-	                            is used instead of the original golden ones
+                                is used instead of the original golden ones
 
 Simon the Sorcerer 1 and 2 add the following non-standard keywords:
 

audio/audiostream.cpp

@@ -386,4 +386,42 @@ Timestamp convertTimeToStreamPos(const Timestamp &where, int rate, bool isStereo
 	return Timestamp(result.secs(), result.numberOfFrames(), result.framerate());
 }
 
+/**
+ * An AudioStream wrapper that cuts off the amount of samples read after a
+ * given time length is reached.
+ */
+class LimitingAudioStream : public AudioStream {
+public:
+	LimitingAudioStream(AudioStream *parentStream, const Audio::Timestamp &length, DisposeAfterUse::Flag disposeAfterUse) :
+			_parentStream(parentStream), _samplesRead(0), _disposeAfterUse(disposeAfterUse),
+			_totalSamples(length.convertToFramerate(getRate()).totalNumberOfFrames() * getChannels()) {}
+
+	~LimitingAudioStream() {
+		if (_disposeAfterUse == DisposeAfterUse::YES)
+			delete _parentStream;
+	}
+
+	int readBuffer(int16 *buffer, const int numSamples) {
+		// Cap us off so we don't read past _totalSamples					
+		int samplesRead = _parentStream->readBuffer(buffer, MIN<int>(numSamples, _totalSamples - _samplesRead));
+		_samplesRead += samplesRead;
+		return samplesRead;
+	}
+
+	bool endOfData() const { return _parentStream->endOfData() || _samplesRead >= _totalSamples; }
+	bool isStereo() const { return _parentStream->isStereo(); }
+	int getRate() const { return _parentStream->getRate(); }
+
+private:
+	int getChannels() const { return isStereo() ? 2 : 1; } 
+
+	AudioStream *_parentStream;
+	DisposeAfterUse::Flag _disposeAfterUse;
+	uint32 _totalSamples, _samplesRead;
+};
+
+AudioStream *makeLimitingAudioStream(AudioStream *parentStream, const Timestamp &length, DisposeAfterUse::Flag disposeAfterUse) {
+	return new LimitingAudioStream(parentStream, length, disposeAfterUse);
+}
+
 } // End of namespace Audio

audio/audiostream.h

@@ -356,6 +356,16 @@ QueuingAudioStream *makeQueuingAudioStream(int rate, bool stereo);
  */
 Timestamp convertTimeToStreamPos(const Timestamp &where, int rate, bool isStereo);
 
+/**
+ * Factory function for an AudioStream wrapper that cuts off the amount of samples read after a
+ * given time length is reached.
+ *
+ * @param parentStream    The stream to limit
+ * @param length          The time length to limit the stream to
+ * @param disposeAfterUse Whether the parent stream object should be destroyed on destruction of the returned stream
+ */
+AudioStream *makeLimitingAudioStream(AudioStream *parentStream, const Timestamp &length, DisposeAfterUse::Flag disposeAfterUse = DisposeAfterUse::YES);
+
 } // End of namespace Audio
 
 #endif

audio/decoders/adpcm.cpp

@@ -71,13 +71,19 @@ int Oki_ADPCMStream::readBuffer(int16 *buffer, const int numSamples) {
 	int samples;
 	byte data;
 
-	assert(numSamples % 2 == 0);
+	for (samples = 0; samples < numSamples && !endOfData(); samples++) {
+		if (_decodedSampleCount == 0) {
+			data = _stream->readByte();
+			_decodedSamples[0] = decodeOKI((data >> 4) & 0x0f);
+			_decodedSamples[1] = decodeOKI((data >> 0) & 0x0f);
+			_decodedSampleCount = 2;
+		}
 
-	for (samples = 0; samples < numSamples && !_stream->eos() && _stream->pos() < _endpos; samples += 2) {
-		data = _stream->readByte();
-		buffer[samples] = decodeOKI((data >> 4) & 0x0f);
-		buffer[samples + 1] = decodeOKI(data & 0x0f);
+		// (1 - (count - 1)) ensures that _decodedSamples acts as a FIFO of depth 2
+		buffer[samples] = _decodedSamples[1 - (_decodedSampleCount - 1)];
+		_decodedSampleCount--;
 	}
+
 	return samples;
 }
 
@@ -117,13 +123,19 @@ int DVI_ADPCMStream::readBuffer(int16 *buffer, const int numSamples) {
 	int samples;
 	byte data;
 
-	assert(numSamples % 2 == 0);
+	for (samples = 0; samples < numSamples && !endOfData(); samples++) {
+		if (_decodedSampleCount == 0) {
+			data = _stream->readByte();
+			_decodedSamples[0] = decodeIMA((data >> 4) & 0x0f, 0);
+			_decodedSamples[1] = decodeIMA((data >> 0) & 0x0f, _channels == 2 ? 1 : 0);
+			_decodedSampleCount = 2;
+		}
 
-	for (samples = 0; samples < numSamples && !_stream->eos() && _stream->pos() < _endpos; samples += 2) {
-		data = _stream->readByte();
-		buffer[samples] = decodeIMA((data >> 4) & 0x0f);
-		buffer[samples + 1] = decodeIMA(data & 0x0f, _channels == 2 ? 1 : 0);
+		// (1 - (count - 1)) ensures that _decodedSamples acts as a FIFO of depth 2
+		buffer[samples] = _decodedSamples[1 - (_decodedSampleCount - 1)];
+		_decodedSampleCount--;
 	}
+
 	return samples;
 }
 

audio/decoders/adpcm_intern.h

@@ -37,7 +37,6 @@
 #include "common/stream.h"
 #include "common/textconsole.h"
 
-
 namespace Audio {
 
 class ADPCMStream : public RewindableAudioStream {
@@ -64,12 +63,11 @@ class ADPCMStream : public RewindableAudioStream {
 	ADPCMStream(Common::SeekableReadStream *stream, DisposeAfterUse::Flag disposeAfterUse, uint32 size, int rate, int channels, uint32 blockAlign);
 
 	virtual bool endOfData() const { return (_stream->eos() || _stream->pos() >= _endpos); }
-	virtual bool isStereo() const	{ return _channels == 2; }
-	virtual int getRate() const	{ return _rate; }
+	virtual bool isStereo() const { return _channels == 2; }
+	virtual int getRate() const { return _rate; }
 
 	virtual bool rewind();
 
-
 	/**
 	 * This table is used by some ADPCM variants (IMA and OKI) to adjust the
 	 * step for use on the next sample.
@@ -83,12 +81,18 @@ class ADPCMStream : public RewindableAudioStream {
 class Oki_ADPCMStream : public ADPCMStream {
 public:
 	Oki_ADPCMStream(Common::SeekableReadStream *stream, DisposeAfterUse::Flag disposeAfterUse, uint32 size, int rate, int channels, uint32 blockAlign)
-		: ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign) {}
+		: ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign) { _decodedSampleCount = 0; }
+
+	virtual bool endOfData() const { return (_stream->eos() || _stream->pos() >= _endpos) && (_decodedSampleCount == 0); }
 
 	virtual int readBuffer(int16 *buffer, const int numSamples);
 
 protected:
 	int16 decodeOKI(byte);
+
+private:
+	uint8 _decodedSampleCount;
+	int16 _decodedSamples[2];
 };
 
 class Ima_ADPCMStream : public ADPCMStream {
@@ -108,9 +112,15 @@ class Ima_ADPCMStream : public ADPCMStream {
 class DVI_ADPCMStream : public Ima_ADPCMStream {
 public:
 	DVI_ADPCMStream(Common::SeekableReadStream *stream, DisposeAfterUse::Flag disposeAfterUse, uint32 size, int rate, int channels, uint32 blockAlign)
-		: Ima_ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign) {}
+		: Ima_ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign) { _decodedSampleCount = 0; }
+
+	virtual bool endOfData() const { return (_stream->eos() || _stream->pos() >= _endpos) && (_decodedSampleCount == 0); }
 
 	virtual int readBuffer(int16 *buffer, const int numSamples);
+
+private:
+	uint8 _decodedSampleCount;
+	int16 _decodedSamples[2];
 };
 
 class Apple_ADPCMStream : public Ima_ADPCMStream {

audio/decoders/qdm2.cpp

@@ -689,7 +689,7 @@ static int getVlc2(Common::BitStream *s, int16 (*table)[2], int bits, int maxDep
 		code = table[index][0];
 		n = table[index][1];
 
-		if(maxDepth > 2 && n < 0) {
+		if (maxDepth > 2 && n < 0) {
 			s->skip(nbBits);
 			index = s->getBits(-n) + code;
 			code = table[index][0];
@@ -861,9 +861,9 @@ void initVlcSparse(VLC *vlc, int nb_bits, int nb_codes,
 		const void *symbols, int symbols_wrap, int symbols_size) {
 	vlc->bits = nb_bits;
 
-	if(vlc->table_size && vlc->table_size == vlc->table_allocated) {
+	if (vlc->table_size && vlc->table_size == vlc->table_allocated) {
 		return;
-	} else if(vlc->table_size) {
+	} else if (vlc->table_size) {
 		error("called on a partially initialized table");
 	}
 
@@ -1353,7 +1353,7 @@ void QDM2Stream::fix_coding_method_array(int sb, int channels, sb_int8_array cod
 
 	for (ch = 0; ch < channels; ch++) {
 		for (j = 0; j < 64; ) {
-			if((coding_method[ch][sb][j] - 8) > 22) {
+			if ((coding_method[ch][sb][j] - 8) > 22) {
 				run = 1;
 				case_val = 8;
 			} else {

audio/decoders/quicktime.cpp

@@ -61,41 +61,6 @@ class SilentAudioStream : public AudioStream {
 	bool _isStereo;
 };
 
-/**
- * An AudioStream wrapper that cuts off the amount of samples read after a
- * given time length is reached.
- */
-class LimitingAudioStream : public AudioStream {
-public:
-	LimitingAudioStream(AudioStream *parentStream, const Audio::Timestamp &length,
-			DisposeAfterUse::Flag disposeAfterUse = DisposeAfterUse::YES) :
-			_parentStream(parentStream), _samplesRead(0), _disposeAfterUse(disposeAfterUse),
-			_totalSamples(length.convertToFramerate(getRate()).totalNumberOfFrames() * getChannels()) {}
-
-	~LimitingAudioStream() {
-		if (_disposeAfterUse == DisposeAfterUse::YES)
-			delete _parentStream;
-	}
-
-	int readBuffer(int16 *buffer, const int numSamples) {
-		// Cap us off so we don't read past _totalSamples					
-		int samplesRead = _parentStream->readBuffer(buffer, MIN<int>(numSamples, _totalSamples - _samplesRead));
-		_samplesRead += samplesRead;
-		return samplesRead;
-	}
-
-	bool endOfData() const { return _parentStream->endOfData() || _samplesRead >= _totalSamples; }
-	bool isStereo() const { return _parentStream->isStereo(); }
-	int getRate() const { return _parentStream->getRate(); }
-
-private:
-	int getChannels() const { return isStereo() ? 2 : 1; } 
-
-	AudioStream *_parentStream;
-	DisposeAfterUse::Flag _disposeAfterUse;
-	uint32 _totalSamples, _samplesRead;
-};
-
 /**
  * An AudioStream wrapper that forces audio to be played in mono.
  * It currently just ignores the right channel if stereo.
@@ -263,7 +228,7 @@ void QuickTimeAudioDecoder::QuickTimeAudioTrack::queueAudio(const Timestamp &len
 				_skipSamples = Timestamp();
 			}
 
-			queueStream(new LimitingAudioStream(new SilentAudioStream(getRate(), isStereo()), editLength), editLength);
+			queueStream(makeLimitingAudioStream(new SilentAudioStream(getRate(), isStereo()), editLength), editLength);
 			_curEdit++;
 			enterNewEdit(nextEditTime);
 		} else {
@@ -289,7 +254,7 @@ void QuickTimeAudioDecoder::QuickTimeAudioTrack::queueAudio(const Timestamp &len
 			// we move on to the next edit
 			if (trackPosition >= nextEditTime || _curChunk >= _parentTrack->chunkCount) {
 				chunkLength = nextEditTime.convertToFramerate(getRate()) - getCurrentTrackTime();
-				stream = new LimitingAudioStream(stream, chunkLength);
+				stream = makeLimitingAudioStream(stream, chunkLength);
 				_curEdit++;
 				enterNewEdit(nextEditTime);
 

audio/mixer.h

@@ -97,7 +97,7 @@ class Mixer : Common::NonCopyable {
 	 * @param stream	the actual AudioStream to be played
 	 * @param id	a unique id assigned to this stream
 	 * @param volume	the volume with which to play the sound, ranging from 0 to 255
-	 * @param balance	the balance with which to play the sound, ranging from -128 to 127
+	 * @param balance	the balance with which to play the sound, ranging from -127 to 127 (full left to full right), 0 is balanced, -128 is invalid
 	 * @param autofreeStream	a flag indicating whether the stream should be
 	 *                          freed after playback finished
 	 * @param permanent	a flag indicating whether a plain stopAll call should