renamed AudioInputStream -> AudioStream

svn-id: r12110

14 files changed, 61 insertions, 61 deletions

diff --git a/sound/audiostream.cpp b/sound/audiostream.cpp
index 457f2b5bc1..2b5c56d971 100644
--- a/sound/audiostream.cpp
+++ b/sound/audiostream.cpp
@@ -54,7 +54,7 @@
  * generated.
  */
 template<bool stereo, bool is16Bit, bool isUnsigned, bool isLE>
-class LinearMemoryStream : public AudioInputStream {
+class LinearMemoryStream : public AudioStream {
 protected:
 	const byte *_ptr;
 	const byte *_end;
@@ -125,7 +125,7 @@ int LinearMemoryStream<stereo, is16Bit, isUnsigned, isLE>::readBuffer(int16 *buf
 
 
 #pragma mark -
-#pragma mark --- WrappedMemoryStream ---
+#pragma mark --- AppendableMemoryStream ---
 #pragma mark -
 
 
@@ -133,7 +133,7 @@ int LinearMemoryStream<stereo, is16Bit, isUnsigned, isLE>::readBuffer(int16 *buf
  * Wrapped memory stream.
  */
 template<bool stereo, bool is16Bit, bool isUnsigned>
-class WrappedMemoryStream : public WrappedAudioInputStream {
+class AppendableMemoryStream : public AppendableAudioStream {
 protected:
 	byte *_bufferStart;
 	byte *_bufferEnd;
@@ -144,8 +144,8 @@ protected:
 
 	inline bool eosIntern() const { return _end == _pos; };
 public:
-	WrappedMemoryStream(int rate, uint bufferSize);
-	~WrappedMemoryStream()		{ free(_bufferStart); }
+	AppendableMemoryStream(int rate, uint bufferSize);
+	~AppendableMemoryStream()		{ free(_bufferStart); }
 	int readBuffer(int16 *buffer, const int numSamples);
 
 	int16 read();
@@ -161,7 +161,7 @@ public:
 
 
 template<bool stereo, bool is16Bit, bool isUnsigned>
-WrappedMemoryStream<stereo, is16Bit, isUnsigned>::WrappedMemoryStream(int rate, uint bufferSize)
+AppendableMemoryStream<stereo, is16Bit, isUnsigned>::AppendableMemoryStream(int rate, uint bufferSize)
  : _finalized(false), _rate(rate) {
 
 	// Verify the buffer size is sane
@@ -176,7 +176,7 @@ WrappedMemoryStream<stereo, is16Bit, isUnsigned>::WrappedMemoryStream(int rate,
 }
 
 template<bool stereo, bool is16Bit, bool isUnsigned>
-inline int16 WrappedMemoryStream<stereo, is16Bit, isUnsigned>::read() {
+inline int16 AppendableMemoryStream<stereo, is16Bit, isUnsigned>::read() {
 	assert(!eosIntern());
 
 	// Wrap around?
@@ -190,7 +190,7 @@ inline int16 WrappedMemoryStream<stereo, is16Bit, isUnsigned>::read() {
 }
 
 template<bool stereo, bool is16Bit, bool isUnsigned>
-int WrappedMemoryStream<stereo, is16Bit, isUnsigned>::readBuffer(int16 *buffer, const int numSamples) {
+int AppendableMemoryStream<stereo, is16Bit, isUnsigned>::readBuffer(int16 *buffer, const int numSamples) {
 	int samples = 0;
 	while (samples < numSamples && !eosIntern()) {
 		// Wrap around?
@@ -209,7 +209,7 @@ int WrappedMemoryStream<stereo, is16Bit, isUnsigned>::readBuffer(int16 *buffer,
 }
 
 template<bool stereo, bool is16Bit, bool isUnsigned>
-void WrappedMemoryStream<stereo, is16Bit, isUnsigned>::append(const byte *data, uint32 len) {
+void AppendableMemoryStream<stereo, is16Bit, isUnsigned>::append(const byte *data, uint32 len) {
 
 	// Verify the buffer size is sane
 	if (is16Bit && stereo)
@@ -225,7 +225,7 @@ void WrappedMemoryStream<stereo, is16Bit, isUnsigned>::append(const byte *data,
 		uint32 size_to_end_of_buffer = _bufferEnd - _end;
 		len -= size_to_end_of_buffer;
 		if ((_end < _pos) || (_bufferStart + len >= _pos)) {
-			warning("WrappedMemoryStream: buffer overflow (A)");
+			warning("AppendableMemoryStream: buffer overflow (A)");
 			return;
 		}
 		memcpy(_end, data, size_to_end_of_buffer);
@@ -233,7 +233,7 @@ void WrappedMemoryStream<stereo, is16Bit, isUnsigned>::append(const byte *data,
 		_end = _bufferStart + len;
 	} else {
 		if ((_end < _pos) && (_end + len >= _pos)) {
-			warning("WrappedMemoryStream: buffer overflow (B)");
+			warning("AppendableMemoryStream: buffer overflow (B)");
 			return;
 		}
 		memcpy(_end, data, len);
@@ -250,7 +250,7 @@ void WrappedMemoryStream<stereo, is16Bit, isUnsigned>::append(const byte *data,
 #if 0
 // Work in progress!!! Not yet usable/finished/working/anything :-)
 
-class ProcInputStream : public AudioInputStream {
+class ProcInputStream : public AudioStream {
 public:
 	typedef void InputProc (void *refCon, int16 *data, uint len);
 
@@ -302,7 +302,7 @@ public:
 		} else \
 			return new LinearMemoryStream<STEREO, false, UNSIGNED, false>(rate, ptr, len, loopOffset, loopLen, autoFree)
 
-AudioInputStream *makeLinearInputStream(int rate, byte _flags, const byte *ptr, uint32 len, uint loopOffset, uint loopLen) {
+AudioStream *makeLinearInputStream(int rate, byte _flags, const byte *ptr, uint32 len, uint loopOffset, uint loopLen) {
 	const bool isStereo   = (_flags & SoundMixer::FLAG_STEREO) != 0;
 	const bool is16Bit    = (_flags & SoundMixer::FLAG_16BITS) != 0;
 	const bool isUnsigned = (_flags & SoundMixer::FLAG_UNSIGNED) != 0;
@@ -326,11 +326,11 @@ AudioInputStream *makeLinearInputStream(int rate, byte _flags, const byte *ptr,
 
 #define MAKE_WRAPPED(STEREO, UNSIGNED) \
 		if (is16Bit) \
-			return new WrappedMemoryStream<STEREO, true, UNSIGNED>(rate, len); \
+			return new AppendableMemoryStream<STEREO, true, UNSIGNED>(rate, len); \
 		else \
-			return new WrappedMemoryStream<STEREO, false, UNSIGNED>(rate, len)
+			return new AppendableMemoryStream<STEREO, false, UNSIGNED>(rate, len)
 
-WrappedAudioInputStream *makeWrappedInputStream(int rate, byte _flags, uint32 len) {
+AppendableAudioStream *makeAppendableAudioStream(int rate, byte _flags, uint32 len) {
 	const bool isStereo = (_flags & SoundMixer::FLAG_STEREO) != 0;
 	const bool is16Bit = (_flags & SoundMixer::FLAG_16BITS) != 0;
 	const bool isUnsigned = (_flags & SoundMixer::FLAG_UNSIGNED) != 0;
diff --git a/sound/audiostream.h b/sound/audiostream.h
index e31c3ac505..ba381880e7 100644
--- a/sound/audiostream.h
+++ b/sound/audiostream.h
@@ -30,9 +30,9 @@
 /**
  * Generic input stream for the resampling code.
  */
-class AudioInputStream {
+class AudioStream {
 public:
-	virtual ~AudioInputStream() {}
+	virtual ~AudioStream() {}
 
 	/**
 	 * Fill the given buffer with up to numSamples samples.
@@ -48,7 +48,7 @@ public:
 	/**
 	 * Read a single (16 bit signed) sample from the stream.
 	 */
-	virtual int16 read() = 0;
+//	virtual int16 read() = 0;
 	
 	/** Is this a stereo stream? */
 	virtual bool isStereo() const = 0;
@@ -76,13 +76,13 @@ public:
 	virtual int getRate() const = 0;
 };
 
-class WrappedAudioInputStream : public AudioInputStream {
+class AppendableAudioStream : public AudioStream {
 public:
 	virtual void append(const byte *data, uint32 len) = 0;
 	virtual void finish() = 0;
 };
 
-class ZeroInputStream : public AudioInputStream {
+class ZeroInputStream : public AudioStream {
 private:
 	int _len;
 public:
@@ -100,7 +100,7 @@ public:
 	int getRate() const { return -1; }
 };
 
-AudioInputStream *makeLinearInputStream(int rate, byte _flags, const byte *ptr, uint32 len, uint loopOffset, uint loopLen);
-WrappedAudioInputStream *makeWrappedInputStream(int rate, byte _flags, uint32 len);
+AudioStream *makeLinearInputStream(int rate, byte _flags, const byte *ptr, uint32 len, uint loopOffset, uint loopLen);
+AppendableAudioStream *makeAppendableAudioStream(int rate, byte _flags, uint32 len);
 
 #endif
diff --git a/sound/mixer.cpp b/sound/mixer.cpp
index 1312e11c38..ff414e46c5 100644
--- a/sound/mixer.cpp
+++ b/sound/mixer.cpp
@@ -52,12 +52,12 @@ private:
 
 protected:
 	RateConverter *_converter;
-	AudioInputStream *_input;
+	AudioStream *_input;
 
 public:
 
 	Channel(SoundMixer *mixer, PlayingSoundHandle *handle, bool isMusic, byte volume, int8 pan, int id = -1);
-	Channel(SoundMixer *mixer, PlayingSoundHandle *handle, AudioInputStream *input, bool autofreeStream, bool isMusic, byte volume, int8 pan, bool reverseStereo = false, int id = -1);
+	Channel(SoundMixer *mixer, PlayingSoundHandle *handle, AudioStream *input, bool autofreeStream, bool isMusic, byte volume, int8 pan, bool reverseStereo = false, int id = -1);
 	virtual ~Channel();
 
 	void mix(int16 *data, uint len);
@@ -229,7 +229,7 @@ void SoundMixer::playRaw(PlayingSoundHandle *handle, void *sound, uint32 size, u
 	}
 
 	// Create the input stream
-	AudioInputStream *input;
+	AudioStream *input;
 	if (flags & SoundMixer::FLAG_LOOP) {
 		if (loopEnd == 0) {
 			input = makeLinearInputStream(rate, flags, (byte *)sound, size, 0, size);
@@ -249,7 +249,7 @@ void SoundMixer::playRaw(PlayingSoundHandle *handle, void *sound, uint32 size, u
 #ifdef USE_MAD
 void SoundMixer::playMP3(PlayingSoundHandle *handle, File *file, uint32 size, byte volume, int8 pan, int id) {
 	// Create the input stream
-	AudioInputStream *input = makeMP3Stream(file, size);
+	AudioStream *input = makeMP3Stream(file, size);
 	playInputStream(handle, input, false, volume, pan, id);
 }
 #endif
@@ -257,12 +257,12 @@ void SoundMixer::playMP3(PlayingSoundHandle *handle, File *file, uint32 size, by
 #ifdef USE_VORBIS
 void SoundMixer::playVorbis(PlayingSoundHandle *handle, File *file, uint32 size, byte volume, int8 pan, int id) {
 	// Create the input stream
-	AudioInputStream *input = makeVorbisStream(file, size);
+	AudioStream *input = makeVorbisStream(file, size);
 	playInputStream(handle, input, false, volume, pan, id);
 }
 #endif
 
-void SoundMixer::playInputStream(PlayingSoundHandle *handle, AudioInputStream *input, bool isMusic, byte volume, int8 pan, int id, bool autofreeStream) {
+void SoundMixer::playInputStream(PlayingSoundHandle *handle, AudioStream *input, bool isMusic, byte volume, int8 pan, int id, bool autofreeStream) {
 	Common::StackLock lock(_mutex);
 
 	if (input == 0) {
@@ -467,7 +467,7 @@ Channel::Channel(SoundMixer *mixer, PlayingSoundHandle *handle, bool isMusic,
 	assert(mixer);
 }
 
-Channel::Channel(SoundMixer *mixer, PlayingSoundHandle *handle, AudioInputStream *input,
+Channel::Channel(SoundMixer *mixer, PlayingSoundHandle *handle, AudioStream *input,
 				bool autofreeStream, bool isMusic, byte volume, int8 pan, bool reverseStereo, int id)
 	: _mixer(mixer), _handle(handle), _autofreeStream(autofreeStream), _isMusic(isMusic),
 	  _volume(volume), _pan(pan), _paused(false), _id(id), _converter(0), _input(input) {
@@ -519,16 +519,16 @@ ChannelStream::ChannelStream(SoundMixer *mixer, PlayingSoundHandle *handle,
 							uint rate, byte flags, uint32 buffer_size, byte volume, int8 pan)
 	: Channel(mixer, handle, true, volume, pan) {
 	// Create the input stream
-	_input = makeWrappedInputStream(rate, flags, buffer_size);
+	_input = makeAppendableAudioStream(rate, flags, buffer_size);
 
 	// Get a rate converter instance
 	_converter = makeRateConverter(_input->getRate(), mixer->getOutputRate(), _input->isStereo(), (flags & SoundMixer::FLAG_REVERSE_STEREO) != 0);
 }
 
 void ChannelStream::finish() {
-	((WrappedAudioInputStream *)_input)->finish();
+	((AppendableAudioStream *)_input)->finish();
 }
 
 void ChannelStream::append(void *data, uint32 len) {
-	((WrappedAudioInputStream *)_input)->append((const byte *)data, len);
+	((AppendableAudioStream *)_input)->append((const byte *)data, len);
 }
diff --git a/sound/mixer.h b/sound/mixer.h
index 6e88b11069..497191f5ee 100644
--- a/sound/mixer.h
+++ b/sound/mixer.h
@@ -28,7 +28,7 @@
 #include "common/system.h"
 
 
-class AudioInputStream;
+class AudioStream;
 class Channel;
 class File;
 
@@ -113,7 +113,7 @@ public:
 	void playVorbis(PlayingSoundHandle *handle, File *file, uint32 size, byte volume = 255, int8 pan = 0, int id = -1);
 #endif
 
-	void playInputStream(PlayingSoundHandle *handle, AudioInputStream *input, bool isMusic, byte volume = 255, int8 pan = 0, int id = -1, bool autofreeStream = true);
+	void playInputStream(PlayingSoundHandle *handle, AudioStream *input, bool isMusic, byte volume = 255, int8 pan = 0, int id = -1, bool autofreeStream = true);
 
 
 	/** Start a new stream. */
diff --git a/sound/mp3.cpp b/sound/mp3.cpp
index 6c74c27d7f..1784300aae 100644
--- a/sound/mp3.cpp
+++ b/sound/mp3.cpp
@@ -37,7 +37,7 @@
 #pragma mark -
 
 
-class MP3InputStream : public AudioInputStream {
+class MP3InputStream : public AudioStream {
 	struct mad_stream _stream;
 	struct mad_frame _frame;
 	struct mad_synth _synth;
@@ -261,7 +261,7 @@ int MP3InputStream::readBuffer(int16 *buffer, const int numSamples) {
 	return samples;
 }
 
-AudioInputStream *makeMP3Stream(File *file, uint size) {
+AudioStream *makeMP3Stream(File *file, uint size) {
 	return new MP3InputStream(file, mad_timer_zero, size);
 }
 
@@ -376,7 +376,7 @@ void MP3TrackInfo::play(SoundMixer *mixer, PlayingSoundHandle *handle, int start
 	}
 
 	// Play it
-	AudioInputStream *input = new MP3InputStream(_file, durationTime, 0);
+	AudioStream *input = new MP3InputStream(_file, durationTime, 0);
 	mixer->playInputStream(handle, input, true);
 }
 
diff --git a/sound/mp3.h b/sound/mp3.h
index b0a1ee323d..e6b11b26a6 100644
--- a/sound/mp3.h
+++ b/sound/mp3.h
@@ -27,13 +27,13 @@
 
 #ifdef USE_MAD
 
-class AudioInputStream;
+class AudioStream;
 class DigitalTrackInfo;
 class File;
 
 DigitalTrackInfo *makeMP3TrackInfo(File *file);
 
-AudioInputStream *makeMP3Stream(File *file, uint size);
+AudioStream *makeMP3Stream(File *file, uint size);
 
 #endif
 
diff --git a/sound/rate.cpp b/sound/rate.cpp
index 352243bc79..5d663867c2 100644
--- a/sound/rate.cpp
+++ b/sound/rate.cpp
@@ -80,7 +80,7 @@ protected:
 
 public:
 	LinearRateConverter(st_rate_t inrate, st_rate_t outrate);
-	int flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r);
+	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);
 	}
@@ -124,7 +124,7 @@ LinearRateConverter<stereo, reverseStereo>::LinearRateConverter(st_rate_t inrate
  * Return number of samples processed.
  */
 template<bool stereo, bool reverseStereo>
-int LinearRateConverter<stereo, reverseStereo>::flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r) {
+int LinearRateConverter<stereo, reverseStereo>::flow(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r) {
 	st_sample_t *ostart, *oend;
 	st_sample_t out[2];
 
@@ -203,7 +203,7 @@ public:
 		free(_buffer);
 	}
 
-	virtual int flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r) {
+	virtual int flow(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r) {
 		assert(input.isStereo() == stereo);
 		
 		st_sample_t *ptr;
diff --git a/sound/rate.h b/sound/rate.h
index f6d0304dcd..bd1c104ab1 100644
--- a/sound/rate.h
+++ b/sound/rate.h
@@ -27,7 +27,7 @@
 #include "common/util.h"
 
 //#include "sound/audiostream.h"
-class AudioInputStream;
+class AudioStream;
 
 typedef int16 st_sample_t;
 typedef uint16 st_volume_t;
@@ -71,7 +71,7 @@ class RateConverter {
 public:
 	RateConverter() {}
 	virtual ~RateConverter() {}
-	virtual int flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r) = 0;
+	virtual int flow(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r) = 0;
 	virtual int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) = 0;
 };
 
diff --git a/sound/resample.cpp b/sound/resample.cpp
index 5a772389fb..fc87619e03 100644
--- a/sound/resample.cpp
+++ b/sound/resample.cpp
@@ -681,7 +681,7 @@ int st_resample_start(resample_t r, st_rate_t inrate, st_rate_t outrate) {
  * Processed signed long samples from ibuf to obuf.
  * Return number of samples processed.
  */
-int st_resample_flow(resample_t r, AudioInputStream &input, st_sample_t *obuf, st_size_t *osamp, st_volume_t vol) {
+int st_resample_flow(resample_t r, AudioStream &input, st_sample_t *obuf, st_size_t *osamp, st_volume_t vol) {
 	long i, k, last;
 	long Nout = 0;	// The number of bytes we effectively output
 	long Nx;		// The number of bytes we will read from input
@@ -689,7 +689,7 @@ int st_resample_flow(resample_t r, AudioInputStream &input, st_sample_t *obuf, s
 	const long obufSize = *osamp;
 
 /*
-TODO: adjust for the changes made to AudioInputStream; add support for stereo
+TODO: adjust for the changes made to AudioStream; add support for stereo
 initially, could just average the left/right channel -> bad for quality of course,
 but easiest to implement and would get this going again.
 Next step is to duplicate the X/Y buffers... a lot of computations don't care about
@@ -946,7 +946,7 @@ ResampleRateConverter::~ResampleRateConverter() {
 	free(Y2);
 }
 
-int ResampleRateConverter::flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol) {
+int ResampleRateConverter::flow(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol) {
 //	return st_resample_flow(&rstuff, input, obuf, &osamp, vol);
 	return 0;
 }
diff --git a/sound/resample.h b/sound/resample.h
index 46c33244ce..e235b95d78 100644
--- a/sound/resample.h
+++ b/sound/resample.h
@@ -85,7 +85,7 @@ protected:
 public:
 	ResampleRateConverter(st_rate_t inrate, st_rate_t outrate, int quality);
 	~ResampleRateConverter();
-	virtual int flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol);
+	virtual int flow(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol);
 	virtual int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol);
 };
 
diff --git a/sound/voc.cpp b/sound/voc.cpp
index f3d3d1319a..f6583b3631 100644
--- a/sound/voc.cpp
+++ b/sound/voc.cpp
@@ -152,14 +152,14 @@ byte *loadVOCFile(File *file, int &size, int &rate) {
 	return data;
 }
 
-AudioInputStream *makeVOCStream(byte *ptr) {
+AudioStream *makeVOCStream(byte *ptr) {
 	int size, rate, loops;
 	byte *data = readVOCFromMemory(ptr, size, rate, loops);
 
 	return makeLinearInputStream(rate, SoundMixer::FLAG_AUTOFREE | SoundMixer::FLAG_UNSIGNED, data, size, 0, 0);
 }
 
-AudioInputStream *makeVOCStream(File *file) {
+AudioStream *makeVOCStream(File *file) {
 	int size, rate;
 	byte *data = loadVOCFile(file, size, rate);
 
diff --git a/sound/voc.h b/sound/voc.h
index fd9da916e5..a57c2a06be 100644
--- a/sound/voc.h
+++ b/sound/voc.h
@@ -26,7 +26,7 @@
 #include "stdafx.h"
 #include "common/scummsys.h"
 
-class AudioInputStream;
+class AudioStream;
 class File;
 
 #if !defined(__GNUC__)
@@ -60,7 +60,7 @@ extern int getSampleRateFromVOCRate(int vocSR);
 extern byte *readVOCFromMemory(byte *ptr, int &size, int &rate, int &loops);
 extern byte *loadVOCFile(File *file, int &size, int &rate);
 
-AudioInputStream *makeVOCStream(byte *ptr);
-AudioInputStream *makeVOCStream(File *file);
+AudioStream *makeVOCStream(byte *ptr);
+AudioStream *makeVOCStream(File *file);
 
 #endif
diff --git a/sound/vorbis.cpp b/sound/vorbis.cpp
index bcee8b792b..e0b7bd0b57 100644
--- a/sound/vorbis.cpp
+++ b/sound/vorbis.cpp
@@ -32,7 +32,7 @@
 #include <vorbis/vorbisfile.h>
 
 
-AudioInputStream *makeVorbisStream(OggVorbis_File *file, int duration);
+AudioStream *makeVorbisStream(OggVorbis_File *file, int duration);
 
 
 #pragma mark -
@@ -145,7 +145,7 @@ void VorbisTrackInfo::play(SoundMixer *mixer, PlayingSoundHandle *handle, int st
 	ov_time_seek(&_ov_file, startFrame / 75.0);
 #endif
 
-	AudioInputStream *input = makeVorbisStream(&_ov_file, duration * ov_info(&_ov_file, -1)->rate / 75);
+	AudioStream *input = makeVorbisStream(&_ov_file, duration * ov_info(&_ov_file, -1)->rate / 75);
 	mixer->playInputStream(handle, input, true);
 }
 
@@ -165,7 +165,7 @@ DigitalTrackInfo *makeVorbisTrackInfo(File *file) {
 #pragma mark -
 
 
-class VorbisInputStream : public AudioInputStream {
+class VorbisInputStream : public AudioStream {
 	OggVorbis_File *_ov_file;
 	int _end_pos;
 	int _numChannels;
@@ -274,11 +274,11 @@ void VorbisInputStream::refill() {
 	_bufferEnd = (int16 *)read_pos;
 }
 
-AudioInputStream *makeVorbisStream(OggVorbis_File *file, int duration) {
+AudioStream *makeVorbisStream(OggVorbis_File *file, int duration) {
 	return new VorbisInputStream(file, duration);
 }
 
-AudioInputStream *makeVorbisStream(File *file, uint32 size) {
+AudioStream *makeVorbisStream(File *file, uint32 size) {
 	OggVorbis_File *ov_file = new OggVorbis_File;
 	file_info *f = new file_info;
 
diff --git a/sound/vorbis.h b/sound/vorbis.h
index 3241f884ae..c09baca3e9 100644
--- a/sound/vorbis.h
+++ b/sound/vorbis.h
@@ -27,13 +27,13 @@
 
 #ifdef USE_VORBIS
 
-class AudioInputStream;
+class AudioStream;
 class DigitalTrackInfo;
 class File;
 
 DigitalTrackInfo *makeVorbisTrackInfo(File *file);
 
-AudioInputStream *makeVorbisStream(File *file, uint32 size);
+AudioStream *makeVorbisStream(File *file, uint32 size);
 
 #endif