Merge branch 'master' into teenagentRefactor

Conflicts:
	engines/teenagent/callbacks.cpp

NEWS

@@ -1,7 +1,26 @@
 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.
+   - Added Galician translation.
+
+  Cine:
+   - Improved audio support for Amiga and AtariST versions of Future Wars.
+     Now music fades out slowly instead of stopping immediately. Sound
+     effects are now properly panned, when requested by the game.
+
+  SCUMM:
+   - Implemented Monkey Island 2 Macintosh's audio driver. Now we properly
+     support its sample based audio output. The same output is also used for
+     the m68k Macintosh version of Indiana Jones and the Fate of Atlantis.
+
+1.5.0 (2012-07-27)
  New Games:
    - Added support for Backyard Baseball 2003.
    - Added support for Blue Force.

README

@@ -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/aiff.h

@@ -23,6 +23,7 @@
 /**
  * @file
  * Sound decoder used in engines:
+ *  - pegasus
  *  - saga
  *  - sci
  *  - sword1

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.
@@ -230,7 +195,7 @@ QuickTimeAudioDecoder::QuickTimeAudioTrack::QuickTimeAudioTrack(QuickTimeAudioDe
 
 	if (entry->getCodecTag() == MKTAG('r', 'a', 'w', ' ') || entry->getCodecTag() == MKTAG('t', 'w', 'o', 's'))
 		_parentTrack->sampleSize = (entry->_bitsPerSample / 8) * entry->_channels;
-	
+
 	// Initialize our edit parser too
 	_curEdit = 0;
 	enterNewEdit(Timestamp());
@@ -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);
 
@@ -430,9 +395,9 @@ AudioStream *QuickTimeAudioDecoder::QuickTimeAudioTrack::readAudioChunk(uint chu
 }
 
 void QuickTimeAudioDecoder::QuickTimeAudioTrack::skipSamples(const Timestamp &length, AudioStream *stream) {
-	uint32 sampleCount = length.convertToFramerate(getRate()).totalNumberOfFrames();
+	int32 sampleCount = length.convertToFramerate(getRate()).totalNumberOfFrames();
 
-	if (sampleCount == 0)
+	if (sampleCount <= 0)
 		return;
 
 	if (isStereo())
@@ -461,7 +426,7 @@ void QuickTimeAudioDecoder::QuickTimeAudioTrack::enterNewEdit(const Timestamp &p
 	// If we're at the end of the edit list, there's nothing else for us to do here
 	if (allDataRead())
 		return;
-	
+
 	// For an empty edit, we may need to adjust the start time
 	if (_parentTrack->editList[_curEdit].mediaTime == -1) {
 		// Just invalidate the current media position (and make sure the scale

audio/decoders/quicktime.h

@@ -25,6 +25,7 @@
  * Sound decoder used in engines:
  *  - groovie
  *  - mohawk
+ *  - pegasus
  *  - sci
  */
 

audio/decoders/quicktime_intern.h

@@ -88,7 +88,7 @@ class QuickTimeAudioDecoder : public Common::QuickTimeParser {
 
 	private:
 		QuickTimeAudioDecoder *_decoder;
-		Track *_parentTrack; 
+		Track *_parentTrack;
 		QueuingAudioStream *_queue;
 		uint _curChunk;
 		Timestamp _curMediaPos, _skipSamples;
@@ -115,7 +115,7 @@ class QuickTimeAudioDecoder : public Common::QuickTimeParser {
 		~AudioSampleDesc();
 
 		bool isAudioCodecSupported() const;
-		
+
 		AudioStream *createAudioStream(Common::SeekableReadStream *stream) const;
 		void initCodec();
 

audio/mididrv.cpp

@@ -240,7 +240,7 @@ MidiDriver::DeviceHandle MidiDriver::detectDevice(int flags) {
 				devStr = ConfMan.hasKey("gm_device") ? ConfMan.get("gm_device") : Common::String("null");
 			else
 				devStr = "auto";
-			
+
 			// Default to Null device here, since we also register a default null setting for
 			// the MT32 or GM device in the config manager.
 			hdl = getDeviceHandle(devStr.empty() ? Common::String("null") : devStr);