Fix invalid sample position on Timestamp to sample conversion for Stereo streams.

svn-id: r47591

6 files changed, 30 insertions, 27 deletions

diff --git a/engines/kyra/sound_digital.cpp b/engines/kyra/sound_digital.cpp
index 393a2f3eda..7a04fc0691 100644
--- a/engines/kyra/sound_digital.cpp
+++ b/engines/kyra/sound_digital.cpp
@@ -356,7 +356,7 @@ int AUDStream::readChunk(int16 *buffer, const int maxSamples) {
 }
 
 bool AUDStream::seek(const Audio::Timestamp &where) {
-	const uint32 seekSample = Audio::calculateSampleOffset(where, getRate());
+	const uint32 seekSample = Audio::convertTimeToStreamPos(where, getRate(), isStereo()).totalNumberOfFrames();
 
 	_stream->seek(_streamStart);
 	_processedSize = 0;
diff --git a/sound/audiostream.cpp b/sound/audiostream.cpp
index e781149a55..08ea636a1d 100644
--- a/sound/audiostream.cpp
+++ b/sound/audiostream.cpp
@@ -164,8 +164,8 @@ SubLoopingAudioStream::SubLoopingAudioStream(SeekableAudioStream *stream,
 	                                         DisposeAfterUse::Flag disposeAfterUse)
     : _parent(stream), _disposeAfterUse(disposeAfterUse), _loops(loops),
       _pos(0, getRate() * (isStereo() ? 2 : 1)),
-      _loopStart(loopStart.convertToFramerate(getRate() * (isStereo() ? 2 : 1))),
-      _loopEnd(loopEnd.convertToFramerate(getRate() * (isStereo() ? 2 : 1))),
+      _loopStart(convertTimeToStreamPos(loopStart, getRate(), isStereo())),
+      _loopEnd(convertTimeToStreamPos(loopEnd, getRate(), isStereo())),
       _done(false) {
 	if (!_parent->rewind())
 		_done = true;
@@ -212,9 +212,9 @@ int SubLoopingAudioStream::readBuffer(int16 *buffer, const int numSamples) {
 
 SubSeekableAudioStream::SubSeekableAudioStream(SeekableAudioStream *parent, const Timestamp start, const Timestamp end, DisposeAfterUse::Flag disposeAfterUse)
     : _parent(parent), _disposeAfterUse(disposeAfterUse),
-      _start(start.convertToFramerate(getRate())),
+      _start(convertTimeToStreamPos(start, getRate(), isStereo())),
        _pos(0, getRate() * (isStereo() ? 2 : 1)),
-      _length((end - start).convertToFramerate(getRate() * (isStereo() ? 2 : 1))) {
+      _length(convertTimeToStreamPos(end - start, getRate(), isStereo())) {
 
 	assert(_length.totalNumberOfFrames() % (isStereo() ? 2 : 1) == 0);
 	_parent->seek(_start);
@@ -233,7 +233,7 @@ int SubSeekableAudioStream::readBuffer(int16 *buffer, const int numSamples) {
 }
 
 bool SubSeekableAudioStream::seek(const Timestamp &where) {
-	_pos = where.convertToFramerate(getRate());
+	_pos = convertTimeToStreamPos(where, getRate(), isStereo());
 	if (_pos > _length) {
 		_pos = _length;
 		return false;
@@ -364,4 +364,15 @@ QueuingAudioStream *makeQueuingAudioStream(int rate, bool stereo) {
 	return new QueuingAudioStreamImpl(rate, stereo);
 }
 
+Timestamp convertTimeToStreamPos(const Timestamp &where, int rate, bool isStereo) {
+	Timestamp result(where.convertToFramerate(rate * (isStereo ? 2 : 1)));
+
+	// When the Stream is a stereo stream, we have to assure
+	// that the sample position is an even number.
+	if (isStereo && (result.totalNumberOfFrames() & 1))
+		return result.addFrames(-1); // We cut off one sample here.
+	else
+		return result;
+}
+
 } // End of namespace Audio
diff --git a/sound/audiostream.h b/sound/audiostream.h
index bb0edaa100..764662e378 100644
--- a/sound/audiostream.h
+++ b/sound/audiostream.h
@@ -162,9 +162,9 @@ public:
 	 * 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 responsibility of the caller.
-	 * @param basename	a filename without an extension
-	 * @return	an SeekableAudioStream ready to use in case of success;
-	 *			NULL in case of an error (e.g. invalid/nonexisting file)
+	 * @param basename  a filename without an extension
+	 * @return  an SeekableAudioStream ready to use in case of success;
+	 *          NULL in case of an error (e.g. invalid/nonexisting file)
 	 */
 	static SeekableAudioStream *openStreamFile(const Common::String &basename);
 
@@ -337,16 +337,15 @@ public:
  */
 QueuingAudioStream *makeQueuingAudioStream(int rate, bool stereo);
 
-
 /**
- * Calculates the sample, which the timestamp describes in a
- * AudioStream with the given framerate.
+ * Converts a point in time to a precise sample offset
+ * with the given parameters.
  *
- * @param where point in time
- * @param rate rate of the AudioStream
- * @return sample index
+ * @param where Point in time.
+ * @param rate Rate of the stream.
+ * @param isStereo Is the stream a stereo stream?
  */
-uint32 calculateSampleOffset(const Timestamp &where, int rate);
+Timestamp convertTimeToStreamPos(const Timestamp &where, int rate, bool isStereo);
 
 } // End of namespace Audio
 
diff --git a/sound/decoders/flac.cpp b/sound/decoders/flac.cpp
index 29302827da..6fa7302a9a 100644
--- a/sound/decoders/flac.cpp
+++ b/sound/decoders/flac.cpp
@@ -294,7 +294,7 @@ bool FlacInputStream::seekAbsolute(FLAC__uint64 sample) {
 bool FlacInputStream::seek(const Timestamp &where) {
 	_sampleCache.bufFill = 0;
 	_sampleCache.bufReadPos = NULL;
-	return seekAbsolute((FLAC__uint64)calculateSampleOffset(where, _streaminfo.sample_rate));
+	return seekAbsolute((FLAC__uint64)convertTimeToStreamPos(where, getRate(), isStereo()).totalNumberOfFrames());
 }
 
 int FlacInputStream::readBuffer(int16 *buffer, const int numSamples) {
diff --git a/sound/decoders/raw.cpp b/sound/decoders/raw.cpp
index 6747111a1a..ecdde10250 100644
--- a/sound/decoders/raw.cpp
+++ b/sound/decoders/raw.cpp
@@ -41,13 +41,6 @@ namespace Audio {
 	((is16Bit ? (isLE ? READ_LE_UINT16(ptr) : READ_BE_UINT16(ptr)) : (*ptr << 8)) ^ (isUnsigned ? 0x8000 : 0))
 
 
-// TODO: Get rid of this
-uint32 calculateSampleOffset(const Timestamp &where, int rate) {
-	return where.convertToFramerate(rate).totalNumberOfFrames();
-}
-
-
-
 #pragma mark -
 #pragma mark --- RawMemoryStream ---
 #pragma mark -
@@ -110,7 +103,7 @@ int RawMemoryStream<stereo, is16Bit, isUnsigned, isLE>::readBuffer(int16 *buffer
 
 template<bool stereo, bool is16Bit, bool isUnsigned, bool isLE>
 bool RawMemoryStream<stereo, is16Bit, isUnsigned, isLE>::seek(const Timestamp &where) {
-	const uint8 *ptr = _origPtr + calculateSampleOffset(where, getRate()) * (is16Bit ? 2 : 1) * (stereo ? 2 : 1);
+	const uint8 *ptr = _origPtr + convertTimeToStreamPos(where, getRate(), isStereo()).totalNumberOfFrames() * (is16Bit ? 2 : 1);
 	if (ptr > _end) {
 		_ptr = _end;
 		return false;
@@ -275,7 +268,7 @@ int RawDiskStream<stereo, is16Bit, isUnsigned, isLE>::readBuffer(int16 *buffer,
 
 template<bool stereo, bool is16Bit, bool isUnsigned, bool isLE>
 bool RawDiskStream<stereo, is16Bit, isUnsigned, isLE>::seek(const Timestamp &where) {
-	const uint32 seekSample = calculateSampleOffset(where, getRate()) * (stereo ? 2 : 1);
+	const uint32 seekSample = convertTimeToStreamPos(where, getRate(), isStereo()).totalNumberOfFrames();
 	uint32 curSample = 0;
 
 	// Search for the disk block in which the specific sample is placed
diff --git a/sound/decoders/vorbis.cpp b/sound/decoders/vorbis.cpp
index 46a94a6214..f5ebc112c3 100644
--- a/sound/decoders/vorbis.cpp
+++ b/sound/decoders/vorbis.cpp
@@ -169,7 +169,7 @@ int VorbisInputStream::readBuffer(int16 *buffer, const int numSamples) {
 }
 
 bool VorbisInputStream::seek(const Timestamp &where) {
-	int res = ov_pcm_seek(&_ovFile, calculateSampleOffset(where, getRate()));
+	int res = ov_pcm_seek(&_ovFile, convertTimeToStreamPos(where, getRate(), isStereo()).totalNumberOfFrames());
 	if (res) {
 		warning("Error seeking in Vorbis stream (%d)", res);
 		_pos = _bufferEnd;