renamed AudioInputStream -> AudioStream
svn-id: r12110
This commit is contained in:
parent
2a6745a986
commit
bfea71b0c4
20 changed files with 72 additions and 72 deletions
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@ -746,7 +746,7 @@ void IMuseDigital::callback() {
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}
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}
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void IMuseDigital::startSound(int sound, byte *voiceBundleData, AudioInputStream *input) {
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void IMuseDigital::startSound(int sound, byte *voiceBundleData, AudioStream *input) {
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debug(5, "IMuseDigital::startSound(%d)", sound);
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int l, r;
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@ -782,7 +782,7 @@ void IMuseDigital::startSound(int sound, byte *voiceBundleData, AudioInputStream
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if (input) {
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// Do nothing here, we already have an audio stream
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} else if (READ_UINT32(ptr) == MKID('Crea')) {
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// Create an AudioInputStream
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// Create an AudioStream
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input = makeVOCStream(ptr);
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} else if (READ_UINT32(ptr) == MKID('iMUS')) {
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uint32 tag;
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@ -922,7 +922,7 @@ void IMuseDigital::startSound(int sound, byte *voiceBundleData, AudioInputStream
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} else
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error("IMuseDigital::startSound(): Can't handle %d bit samples", bits);
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// Create an AudioInputStream
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// Create an AudioStream
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input = makeLinearInputStream(freq, mixerFlags | SoundMixer::FLAG_AUTOFREE, data, size, 0, 0);
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} else {
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error("IMuseDigital::startSound(): Unknown sound format");
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@ -27,7 +27,7 @@
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#include "scumm/music.h"
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#include "sound/mixer.h"
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class AudioInputStream;
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class AudioStream;
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namespace Scumm {
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@ -81,7 +81,7 @@ private:
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bool used;
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bool started;
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PlayingSoundHandle handle;
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AudioInputStream *stream;
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AudioStream *stream;
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Channel();
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};
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@ -128,7 +128,7 @@ public:
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void stopBundleMusic();
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void playBundleSound(const char *sound);
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void startSound(int sound, byte *voiceBundleData, AudioInputStream *input);
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void startSound(int sound, byte *voiceBundleData, AudioStream *input);
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public:
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IMuseDigital(ScummEngine *scumm);
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@ -25,7 +25,7 @@
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#include "scumm/scumm.h"
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class AudioInputStream;
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class AudioStream;
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class RateConverter;
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namespace Scumm {
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@ -61,7 +61,7 @@ private:
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int8 pan;
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uint16 freq;
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RateConverter *converter;
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AudioInputStream *input;
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AudioStream *input;
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};
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SoundMixer *_mixer;
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@ -813,7 +813,7 @@ void Sound::pauseSounds(bool pause) {
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void Sound::startSfxSound(File *file, int file_size, PlayingSoundHandle *handle, int id) {
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AudioInputStream *input;
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AudioStream *input;
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if (file_size > 0) {
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@ -54,7 +54,7 @@
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* generated.
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*/
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template<bool stereo, bool is16Bit, bool isUnsigned, bool isLE>
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class LinearMemoryStream : public AudioInputStream {
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class LinearMemoryStream : public AudioStream {
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protected:
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const byte *_ptr;
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const byte *_end;
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@ -125,7 +125,7 @@ int LinearMemoryStream<stereo, is16Bit, isUnsigned, isLE>::readBuffer(int16 *buf
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#pragma mark -
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#pragma mark --- WrappedMemoryStream ---
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#pragma mark --- AppendableMemoryStream ---
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#pragma mark -
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@ -133,7 +133,7 @@ int LinearMemoryStream<stereo, is16Bit, isUnsigned, isLE>::readBuffer(int16 *buf
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* Wrapped memory stream.
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*/
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template<bool stereo, bool is16Bit, bool isUnsigned>
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class WrappedMemoryStream : public WrappedAudioInputStream {
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class AppendableMemoryStream : public AppendableAudioStream {
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protected:
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byte *_bufferStart;
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byte *_bufferEnd;
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@ -144,8 +144,8 @@ protected:
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inline bool eosIntern() const { return _end == _pos; };
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public:
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WrappedMemoryStream(int rate, uint bufferSize);
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~WrappedMemoryStream() { free(_bufferStart); }
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AppendableMemoryStream(int rate, uint bufferSize);
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~AppendableMemoryStream() { free(_bufferStart); }
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int readBuffer(int16 *buffer, const int numSamples);
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int16 read();
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@ -161,7 +161,7 @@ public:
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template<bool stereo, bool is16Bit, bool isUnsigned>
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WrappedMemoryStream<stereo, is16Bit, isUnsigned>::WrappedMemoryStream(int rate, uint bufferSize)
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AppendableMemoryStream<stereo, is16Bit, isUnsigned>::AppendableMemoryStream(int rate, uint bufferSize)
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: _finalized(false), _rate(rate) {
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// Verify the buffer size is sane
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@ -176,7 +176,7 @@ WrappedMemoryStream<stereo, is16Bit, isUnsigned>::WrappedMemoryStream(int rate,
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}
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template<bool stereo, bool is16Bit, bool isUnsigned>
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inline int16 WrappedMemoryStream<stereo, is16Bit, isUnsigned>::read() {
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inline int16 AppendableMemoryStream<stereo, is16Bit, isUnsigned>::read() {
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assert(!eosIntern());
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// Wrap around?
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@ -190,7 +190,7 @@ inline int16 WrappedMemoryStream<stereo, is16Bit, isUnsigned>::read() {
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}
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template<bool stereo, bool is16Bit, bool isUnsigned>
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int WrappedMemoryStream<stereo, is16Bit, isUnsigned>::readBuffer(int16 *buffer, const int numSamples) {
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int AppendableMemoryStream<stereo, is16Bit, isUnsigned>::readBuffer(int16 *buffer, const int numSamples) {
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int samples = 0;
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while (samples < numSamples && !eosIntern()) {
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// Wrap around?
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@ -209,7 +209,7 @@ int WrappedMemoryStream<stereo, is16Bit, isUnsigned>::readBuffer(int16 *buffer,
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}
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template<bool stereo, bool is16Bit, bool isUnsigned>
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void WrappedMemoryStream<stereo, is16Bit, isUnsigned>::append(const byte *data, uint32 len) {
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void AppendableMemoryStream<stereo, is16Bit, isUnsigned>::append(const byte *data, uint32 len) {
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// Verify the buffer size is sane
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if (is16Bit && stereo)
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@ -225,7 +225,7 @@ void WrappedMemoryStream<stereo, is16Bit, isUnsigned>::append(const byte *data,
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uint32 size_to_end_of_buffer = _bufferEnd - _end;
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len -= size_to_end_of_buffer;
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if ((_end < _pos) || (_bufferStart + len >= _pos)) {
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warning("WrappedMemoryStream: buffer overflow (A)");
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warning("AppendableMemoryStream: buffer overflow (A)");
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return;
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}
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memcpy(_end, data, size_to_end_of_buffer);
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@ -233,7 +233,7 @@ void WrappedMemoryStream<stereo, is16Bit, isUnsigned>::append(const byte *data,
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_end = _bufferStart + len;
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} else {
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if ((_end < _pos) && (_end + len >= _pos)) {
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warning("WrappedMemoryStream: buffer overflow (B)");
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warning("AppendableMemoryStream: buffer overflow (B)");
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return;
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}
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memcpy(_end, data, len);
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@ -250,7 +250,7 @@ void WrappedMemoryStream<stereo, is16Bit, isUnsigned>::append(const byte *data,
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#if 0
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// Work in progress!!! Not yet usable/finished/working/anything :-)
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class ProcInputStream : public AudioInputStream {
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class ProcInputStream : public AudioStream {
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public:
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typedef void InputProc (void *refCon, int16 *data, uint len);
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@ -302,7 +302,7 @@ public:
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} else \
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return new LinearMemoryStream<STEREO, false, UNSIGNED, false>(rate, ptr, len, loopOffset, loopLen, autoFree)
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AudioInputStream *makeLinearInputStream(int rate, byte _flags, const byte *ptr, uint32 len, uint loopOffset, uint loopLen) {
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AudioStream *makeLinearInputStream(int rate, byte _flags, const byte *ptr, uint32 len, uint loopOffset, uint loopLen) {
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const bool isStereo = (_flags & SoundMixer::FLAG_STEREO) != 0;
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const bool is16Bit = (_flags & SoundMixer::FLAG_16BITS) != 0;
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const bool isUnsigned = (_flags & SoundMixer::FLAG_UNSIGNED) != 0;
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@ -326,11 +326,11 @@ AudioInputStream *makeLinearInputStream(int rate, byte _flags, const byte *ptr,
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#define MAKE_WRAPPED(STEREO, UNSIGNED) \
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if (is16Bit) \
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return new WrappedMemoryStream<STEREO, true, UNSIGNED>(rate, len); \
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return new AppendableMemoryStream<STEREO, true, UNSIGNED>(rate, len); \
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else \
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return new WrappedMemoryStream<STEREO, false, UNSIGNED>(rate, len)
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return new AppendableMemoryStream<STEREO, false, UNSIGNED>(rate, len)
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WrappedAudioInputStream *makeWrappedInputStream(int rate, byte _flags, uint32 len) {
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AppendableAudioStream *makeAppendableAudioStream(int rate, byte _flags, uint32 len) {
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const bool isStereo = (_flags & SoundMixer::FLAG_STEREO) != 0;
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const bool is16Bit = (_flags & SoundMixer::FLAG_16BITS) != 0;
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const bool isUnsigned = (_flags & SoundMixer::FLAG_UNSIGNED) != 0;
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@ -30,9 +30,9 @@
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/**
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* Generic input stream for the resampling code.
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*/
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class AudioInputStream {
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class AudioStream {
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public:
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virtual ~AudioInputStream() {}
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virtual ~AudioStream() {}
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/**
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* Fill the given buffer with up to numSamples samples.
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@ -48,7 +48,7 @@ public:
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/**
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* Read a single (16 bit signed) sample from the stream.
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*/
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virtual int16 read() = 0;
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// virtual int16 read() = 0;
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/** Is this a stereo stream? */
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virtual bool isStereo() const = 0;
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virtual int getRate() const = 0;
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};
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class WrappedAudioInputStream : public AudioInputStream {
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class AppendableAudioStream : public AudioStream {
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public:
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virtual void append(const byte *data, uint32 len) = 0;
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virtual void finish() = 0;
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};
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class ZeroInputStream : public AudioInputStream {
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class ZeroInputStream : public AudioStream {
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private:
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int _len;
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public:
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int getRate() const { return -1; }
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};
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AudioInputStream *makeLinearInputStream(int rate, byte _flags, const byte *ptr, uint32 len, uint loopOffset, uint loopLen);
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WrappedAudioInputStream *makeWrappedInputStream(int rate, byte _flags, uint32 len);
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AudioStream *makeLinearInputStream(int rate, byte _flags, const byte *ptr, uint32 len, uint loopOffset, uint loopLen);
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AppendableAudioStream *makeAppendableAudioStream(int rate, byte _flags, uint32 len);
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#endif
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@ -52,12 +52,12 @@ private:
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protected:
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RateConverter *_converter;
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AudioInputStream *_input;
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AudioStream *_input;
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public:
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Channel(SoundMixer *mixer, PlayingSoundHandle *handle, bool isMusic, byte volume, int8 pan, int id = -1);
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Channel(SoundMixer *mixer, PlayingSoundHandle *handle, AudioInputStream *input, bool autofreeStream, bool isMusic, byte volume, int8 pan, bool reverseStereo = false, int id = -1);
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Channel(SoundMixer *mixer, PlayingSoundHandle *handle, AudioStream *input, bool autofreeStream, bool isMusic, byte volume, int8 pan, bool reverseStereo = false, int id = -1);
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virtual ~Channel();
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void mix(int16 *data, uint len);
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@ -229,7 +229,7 @@ void SoundMixer::playRaw(PlayingSoundHandle *handle, void *sound, uint32 size, u
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}
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// Create the input stream
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AudioInputStream *input;
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AudioStream *input;
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if (flags & SoundMixer::FLAG_LOOP) {
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if (loopEnd == 0) {
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input = makeLinearInputStream(rate, flags, (byte *)sound, size, 0, size);
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@ -249,7 +249,7 @@ void SoundMixer::playRaw(PlayingSoundHandle *handle, void *sound, uint32 size, u
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#ifdef USE_MAD
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void SoundMixer::playMP3(PlayingSoundHandle *handle, File *file, uint32 size, byte volume, int8 pan, int id) {
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// Create the input stream
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AudioInputStream *input = makeMP3Stream(file, size);
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AudioStream *input = makeMP3Stream(file, size);
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playInputStream(handle, input, false, volume, pan, id);
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}
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#endif
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@ -257,12 +257,12 @@ void SoundMixer::playMP3(PlayingSoundHandle *handle, File *file, uint32 size, by
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#ifdef USE_VORBIS
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void SoundMixer::playVorbis(PlayingSoundHandle *handle, File *file, uint32 size, byte volume, int8 pan, int id) {
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// Create the input stream
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AudioInputStream *input = makeVorbisStream(file, size);
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AudioStream *input = makeVorbisStream(file, size);
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playInputStream(handle, input, false, volume, pan, id);
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}
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#endif
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void SoundMixer::playInputStream(PlayingSoundHandle *handle, AudioInputStream *input, bool isMusic, byte volume, int8 pan, int id, bool autofreeStream) {
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void SoundMixer::playInputStream(PlayingSoundHandle *handle, AudioStream *input, bool isMusic, byte volume, int8 pan, int id, bool autofreeStream) {
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Common::StackLock lock(_mutex);
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if (input == 0) {
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assert(mixer);
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}
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Channel::Channel(SoundMixer *mixer, PlayingSoundHandle *handle, AudioInputStream *input,
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Channel::Channel(SoundMixer *mixer, PlayingSoundHandle *handle, AudioStream *input,
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bool autofreeStream, bool isMusic, byte volume, int8 pan, bool reverseStereo, int id)
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: _mixer(mixer), _handle(handle), _autofreeStream(autofreeStream), _isMusic(isMusic),
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_volume(volume), _pan(pan), _paused(false), _id(id), _converter(0), _input(input) {
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@ -519,16 +519,16 @@ ChannelStream::ChannelStream(SoundMixer *mixer, PlayingSoundHandle *handle,
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uint rate, byte flags, uint32 buffer_size, byte volume, int8 pan)
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: Channel(mixer, handle, true, volume, pan) {
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// Create the input stream
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_input = makeWrappedInputStream(rate, flags, buffer_size);
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_input = makeAppendableAudioStream(rate, flags, buffer_size);
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// Get a rate converter instance
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_converter = makeRateConverter(_input->getRate(), mixer->getOutputRate(), _input->isStereo(), (flags & SoundMixer::FLAG_REVERSE_STEREO) != 0);
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}
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void ChannelStream::finish() {
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((WrappedAudioInputStream *)_input)->finish();
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((AppendableAudioStream *)_input)->finish();
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}
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void ChannelStream::append(void *data, uint32 len) {
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((WrappedAudioInputStream *)_input)->append((const byte *)data, len);
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((AppendableAudioStream *)_input)->append((const byte *)data, len);
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}
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@ -28,7 +28,7 @@
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#include "common/system.h"
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class AudioInputStream;
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class AudioStream;
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class Channel;
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class File;
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void playVorbis(PlayingSoundHandle *handle, File *file, uint32 size, byte volume = 255, int8 pan = 0, int id = -1);
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#endif
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void playInputStream(PlayingSoundHandle *handle, AudioInputStream *input, bool isMusic, byte volume = 255, int8 pan = 0, int id = -1, bool autofreeStream = true);
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void playInputStream(PlayingSoundHandle *handle, AudioStream *input, bool isMusic, byte volume = 255, int8 pan = 0, int id = -1, bool autofreeStream = true);
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/** Start a new stream. */
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#pragma mark -
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class MP3InputStream : public AudioInputStream {
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class MP3InputStream : public AudioStream {
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struct mad_stream _stream;
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struct mad_frame _frame;
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struct mad_synth _synth;
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@ -261,7 +261,7 @@ int MP3InputStream::readBuffer(int16 *buffer, const int numSamples) {
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return samples;
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}
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AudioInputStream *makeMP3Stream(File *file, uint size) {
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AudioStream *makeMP3Stream(File *file, uint size) {
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return new MP3InputStream(file, mad_timer_zero, size);
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}
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@ -376,7 +376,7 @@ void MP3TrackInfo::play(SoundMixer *mixer, PlayingSoundHandle *handle, int start
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}
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// Play it
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AudioInputStream *input = new MP3InputStream(_file, durationTime, 0);
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AudioStream *input = new MP3InputStream(_file, durationTime, 0);
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mixer->playInputStream(handle, input, true);
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}
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@ -27,13 +27,13 @@
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#ifdef USE_MAD
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class AudioInputStream;
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class AudioStream;
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class DigitalTrackInfo;
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class File;
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DigitalTrackInfo *makeMP3TrackInfo(File *file);
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AudioInputStream *makeMP3Stream(File *file, uint size);
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AudioStream *makeMP3Stream(File *file, uint size);
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#endif
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@ -80,7 +80,7 @@ protected:
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public:
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LinearRateConverter(st_rate_t inrate, st_rate_t outrate);
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int flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r);
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int flow(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r);
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int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) {
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return (ST_SUCCESS);
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}
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@ -124,7 +124,7 @@ LinearRateConverter<stereo, reverseStereo>::LinearRateConverter(st_rate_t inrate
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* Return number of samples processed.
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*/
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template<bool stereo, bool reverseStereo>
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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) {
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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) {
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st_sample_t *ostart, *oend;
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st_sample_t out[2];
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free(_buffer);
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}
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|
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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;
|
||||
|
|
|
@ -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;
|
||||
};
|
||||
|
||||
|
|
|
@ -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;
|
||||
}
|
||||
|
|
|
@ -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);
|
||||
};
|
||||
|
||||
|
|
|
@ -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);
|
||||
|
||||
|
|
|
@ -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
|
||||
|
|
|
@ -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;
|
||||
|
||||
|
|
|
@ -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
|
||||
|
||||
|
|
|
@ -34,7 +34,7 @@
|
|||
|
||||
class SoundMixer;
|
||||
|
||||
class SwordMusicHandle : public AudioInputStream {
|
||||
class SwordMusicHandle : public AudioStream {
|
||||
private:
|
||||
File _file;
|
||||
bool _looping;
|
||||
|
|
|
@ -44,7 +44,7 @@ struct FxHandle {
|
|||
PlayingSoundHandle _handle;
|
||||
};
|
||||
|
||||
class MusicHandle : public AudioInputStream {
|
||||
class MusicHandle : public AudioStream {
|
||||
public:
|
||||
bool _firstTime;
|
||||
bool _streaming;
|
||||
|
|
Loading…
Add table
Add a link
Reference in a new issue