From 6b4484472b79dc7ea7d1ce545a28fba7d3b7696f Mon Sep 17 00:00:00 2001 From: Eugene Sandulenko Date: Sat, 30 Jul 2005 21:11:48 +0000 Subject: Remove trailing whitespaces. svn-id: r18604 --- sound/audiocd.cpp | 4 +-- sound/audiostream.cpp | 18 ++++++------- sound/audiostream.h | 8 +++--- sound/flac.cpp | 54 +++++++++++++++++++-------------------- sound/fmopl.cpp | 8 +++--- sound/fmopl.h | 4 +-- sound/mididrv.cpp | 4 +-- sound/mididrv.h | 2 +- sound/midiparser_xmidi.cpp | 8 +++--- sound/mixer.cpp | 10 ++++---- sound/mixer.h | 6 ++--- sound/mp3.cpp | 6 ++--- sound/mpu401.cpp | 2 +- sound/rate.cpp | 8 +++--- sound/softsynth/adlib.cpp | 26 +++++++++---------- sound/softsynth/emumidi.h | 6 ++--- sound/softsynth/fluidsynth.cpp | 2 +- sound/softsynth/mt32.cpp | 12 ++++----- sound/softsynth/mt32/freeverb.cpp | 8 +++--- sound/softsynth/mt32/freeverb.h | 6 ++--- sound/softsynth/mt32/part.h | 2 +- sound/softsynth/mt32/synth.cpp | 14 +++++----- sound/softsynth/mt32/tables.cpp | 2 +- sound/softsynth/ym2612.cpp | 2 +- sound/vorbis.cpp | 12 ++++----- sound/wave.cpp | 12 ++++----- 26 files changed, 123 insertions(+), 123 deletions(-) (limited to 'sound') diff --git a/sound/audiocd.cpp b/sound/audiocd.cpp index ab894724d4..7be50dc4d6 100644 --- a/sound/audiocd.cpp +++ b/sound/audiocd.cpp @@ -33,7 +33,7 @@ struct TrackFormat { /** Decodername */ const char* decoderName; - /** + /** * Pointer to a function which tries to open the specified track - the only argument * is the number of the track to be played. * Returns either the DigitalTrackInfo object representing the requested track or null @@ -43,7 +43,7 @@ struct TrackFormat { }; static const TrackFormat TRACK_FORMATS[] = { - /* decoderName, openTrackFunction */ + /* decoderName, openTrackFunction */ #ifdef USE_FLAC { "Flac", getFlacTrack }, #endif // #ifdef USE_FLAC diff --git a/sound/audiostream.cpp b/sound/audiostream.cpp index e8ac3a4ab3..9cab081f69 100644 --- a/sound/audiostream.cpp +++ b/sound/audiostream.cpp @@ -32,15 +32,15 @@ struct StreamFileFormat { /** Decodername */ const char* decoderName; const char* fileExtension; - /** + /** * Pointer to a function which tries to open a file of type StreamFormat. - * Return NULL in case of an error (invalid/nonexisting file). + * Return NULL in case of an error (invalid/nonexisting file). */ AudioStream* (*openStreamFile)(Common::File *file, uint32 size); }; -static const StreamFileFormat STREAM_FILEFORMATS[] = { - /* decoderName, fileExt, openStreamFuntion */ +static const StreamFileFormat STREAM_FILEFORMATS[] = { + /* decoderName, fileExt, openStreamFuntion */ #ifdef USE_FLAC { "Flac", "flac", makeFlacStream }, { "Flac", "fla", makeFlacStream }, @@ -60,7 +60,7 @@ AudioStream* AudioStream::openStreamFile(const char *filename) char buffer[1024]; const uint len = strlen(filename); assert(len+6 < sizeof(buffer)); // we need a bigger buffer if wrong - + memcpy(buffer, filename, len); buffer[len] = '.'; char *ext = &buffer[len+1]; @@ -74,7 +74,7 @@ AudioStream* AudioStream::openStreamFile(const char *filename) if (fileHandle->isOpen()) stream = STREAM_FILEFORMATS[i].openStreamFile(fileHandle, fileHandle->size()); } - + // Do not reference the file anymore. If the stream didn't incRef the file, // the object will be deleted (and the file be closed). fileHandle->decRef(); @@ -93,7 +93,7 @@ AudioStream* AudioStream::openStreamFile(const char *filename) /** * A simple raw audio stream, purely memory based. It operates on a single - * block of data, which is passed to it upon creation. + * block of data, which is passed to it upon creation. * Optionally supports looping the sound. * * Design note: This code tries to be as optimized as possible (without @@ -129,7 +129,7 @@ public: } if (stereo) // Stereo requires even sized data assert(len % 2 == 0); - + _origPtr = autoFreeMemory ? ptr : 0; } ~LinearMemoryStream() { @@ -191,7 +191,7 @@ AudioStream *makeLinearInputStream(int rate, byte flags, const byte *ptr, uint32 const bool isUnsigned = (flags & Audio::Mixer::FLAG_UNSIGNED) != 0; const bool isLE = (flags & Audio::Mixer::FLAG_LITTLE_ENDIAN) != 0; const bool autoFree = (flags & Audio::Mixer::FLAG_AUTOFREE) != 0; - + if (isStereo) { if (isUnsigned) { MAKE_LINEAR(true, true); diff --git a/sound/audiostream.h b/sound/audiostream.h index 13a70afa65..798c77e6c7 100644 --- a/sound/audiostream.h +++ b/sound/audiostream.h @@ -47,7 +47,7 @@ public: /** Is this a stereo stream? */ virtual bool isStereo() const = 0; - + /** * End of data reached? If this returns true, it means that at this * time there is no data available in the stream. However there may be @@ -56,7 +56,7 @@ public: * converting data or stop. */ virtual bool endOfData() const = 0; - + /** * End of stream reached? If this returns true, it means that all data * in this stream is used up and no additional data will appear in it @@ -75,7 +75,7 @@ public: * In case of an error, the file handle will be closed, but deleting * it is still the responsibilty of the caller. * @param filename a filename without an extension - * @return an Audiostream ready to use in case of success; + * @return an Audiostream ready to use in case of success; * NULL in case of an error (e.g. invalid/nonexisting file) */ static AudioStream* openStreamFile(const char *filename); @@ -98,7 +98,7 @@ public: } bool isStereo() const { return false; } bool eos() const { return _len <= 0; } - + int getRate() const { return -1; } }; diff --git a/sound/flac.cpp b/sound/flac.cpp index c5840476ed..a6551e544d 100644 --- a/sound/flac.cpp +++ b/sound/flac.cpp @@ -86,7 +86,7 @@ protected: inline ::FLAC__StreamDecoderWriteStatus callbackWrite(const ::FLAC__Frame *frame, const FLAC__int32 * const buffer[]); inline void callbackMetadata(const ::FLAC__StreamMetadata *metadata); inline void callbackError(::FLAC__StreamDecoderErrorStatus status); - + ::FLAC__SeekableStreamDecoder *_decoder; private: @@ -103,14 +103,14 @@ private: void operator=(const FlacInputStream &); bool isValid() const { return _decoder != NULL; } - + bool allocateBuffer(uint minSamples); inline void flushBuffer(); inline void deleteBuffer(); - + /** Header of the Stream */ FLAC__StreamMetadata_StreamInfo _streaminfo; - + struct { /** Handle to the File */ File *fileHandle; @@ -121,25 +121,25 @@ private: /** last index of Stream + 1(!) - not necessary end of file */ uint32 fileEndPos; } _fileInfo; - + /** index of the first Sample to be played */ FLAC__uint64 _firstSample; /** index + 1(!) of the last Sample to be played - 0 is end of Stream*/ FLAC__uint64 _lastSample; - + /** true if the last Sample was decoded from the FLAC-API - there might still be data in the buffer */ bool _lastSampleWritten; - + typedef int16 bufType; enum { BUFTYPE_BITS = 16 }; - + struct { bufType *bufData; bufType *bufReadPos; uint bufSize; uint bufFill; } _preBuffer; - + bufType *_outBuffer; uint _requestedSamples; @@ -154,7 +154,7 @@ private: }; FlacInputStream::FlacInputStream(File *sourceFile, const uint32 fileStart) - : _decoder(::FLAC__seekable_stream_decoder_new()), _firstSample(0), _lastSample(0), + : _decoder(::FLAC__seekable_stream_decoder_new()), _firstSample(0), _lastSample(0), _outBuffer(NULL), _requestedSamples(0), _lastSampleWritten(true), _methodConvertBuffers(&FlacInputStream::convertBuffersGeneric) { @@ -170,15 +170,15 @@ FlacInputStream::FlacInputStream(File *sourceFile, const uint32 fileStart) _fileInfo.fileStartPos = fileStart; _fileInfo.filePos = fileStart; _fileInfo.fileEndPos = sourceFile->size(); - + _fileInfo.fileHandle->incRef(); } -FlacInputStream::FlacInputStream(File *sourceFile, const uint32 fileStart, const uint32 fileStop) - : _decoder(::FLAC__seekable_stream_decoder_new()), _firstSample(0), _lastSample(0), +FlacInputStream::FlacInputStream(File *sourceFile, const uint32 fileStart, const uint32 fileStop) + : _decoder(::FLAC__seekable_stream_decoder_new()), _firstSample(0), _lastSample(0), _outBuffer(NULL), _requestedSamples(0), _lastSampleWritten(true), _methodConvertBuffers(&FlacInputStream::convertBuffersGeneric) -{ +{ assert(sourceFile != NULL && sourceFile->isOpen()); assert(fileStop <= 0 || (fileStart < fileStop && fileStop <= sourceFile->size())); @@ -192,7 +192,7 @@ FlacInputStream::FlacInputStream(File *sourceFile, const uint32 fileStart, const _fileInfo.fileStartPos = fileStart; _fileInfo.filePos = fileStart; _fileInfo.fileEndPos = fileStop; - + _fileInfo.fileHandle->incRef(); } @@ -203,7 +203,7 @@ FlacInputStream::~FlacInputStream() { } if (_preBuffer.bufData != NULL) delete[] _preBuffer.bufData; - + _fileInfo.fileHandle->decRef(); } @@ -246,7 +246,7 @@ bool FlacInputStream::init() { } warning("FlacInputStream: could not create an Audiostream from File %s", _fileInfo.fileHandle->name()); - return false; + return false; } bool FlacInputStream::finish() { @@ -304,7 +304,7 @@ int FlacInputStream::readBuffer(int16 *buffer, const int numSamples) { const uint copySamples = MIN((uint)numSamples, _preBuffer.bufFill); memcpy(buffer, _preBuffer.bufReadPos, copySamples*sizeof(buffer[0])); - + _outBuffer = buffer + copySamples; _requestedSamples = numSamples - copySamples; _preBuffer.bufReadPos += copySamples; @@ -351,9 +351,9 @@ inline ::FLAC__SeekableStreamDecoderReadStatus FlacInputStream::callbackRead(FLA if (*bytes == 0) return FLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_ERROR; /* abort to avoid a deadlock */ - + const uint32 length = MIN(_fileInfo.fileEndPos - _fileInfo.filePos, static_cast(*bytes)); - + _fileInfo.fileHandle->seek(_fileInfo.filePos); const uint32 bytesRead = _fileInfo.fileHandle->read(buffer, length); @@ -365,7 +365,7 @@ inline ::FLAC__SeekableStreamDecoderReadStatus FlacInputStream::callbackRead(FLA return FLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_OK; } -inline void FlacInputStream::setLastSample(FLAC__uint64 absoluteSample) { +inline void FlacInputStream::setLastSample(FLAC__uint64 absoluteSample) { if (_lastSampleWritten && absoluteSample > _lastSample) _lastSampleWritten = false; _lastSample = absoluteSample; @@ -555,14 +555,14 @@ void FlacInputStream::convertBuffersGeneric(bufType* bufDestination, const FLAC_ if (numBits < BUFTYPE_BITS) { const uint8 kPower = (uint8)(BUFTYPE_BITS - numBits); - + for (; numSamples > 0; numSamples -= numChannels) { for (uint i = 0; i < numChannels; ++i) *bufDestination++ = static_cast(*(inChannels[i]++)) << kPower; } } else if (numBits > BUFTYPE_BITS) { const uint8 kPower = (uint8)(numBits - BUFTYPE_BITS); - + for (; numSamples > 0; numSamples -= numChannels) { for (uint i = 0; i < numChannels; ++i) *bufDestination++ = static_cast(*(inChannels[i]++) >> kPower) ; @@ -582,7 +582,7 @@ inline ::FLAC__StreamDecoderWriteStatus FlacInputStream::callbackWrite(const ::F assert(frame->header.sample_rate == _streaminfo.sample_rate); assert(frame->header.bits_per_sample == _streaminfo.bits_per_sample); assert(frame->header.number_type == FLAC__FRAME_NUMBER_TYPE_SAMPLE_NUMBER || _streaminfo.min_blocksize == _streaminfo.max_blocksize); - + assert(_preBuffer.bufFill == 0); // we dont append data uint nSamples = frame->header.blocksize; @@ -611,7 +611,7 @@ inline ::FLAC__StreamDecoderWriteStatus FlacInputStream::callbackWrite(const ::F if (_requestedSamples > 0) { assert(_requestedSamples % kNumChannels == 0); // must be integral multiply of channels assert(_outBuffer != NULL); - + const uint copySamples = MIN(_requestedSamples,nSamples); (*_methodConvertBuffers)(_outBuffer, inChannels, copySamples, kNumChannels, kNumBits); @@ -674,7 +674,7 @@ inline void FlacInputStream::callbackMetadata(const ::FLAC__StreamMetadata *meta } inline void FlacInputStream::callbackError(::FLAC__StreamDecoderErrorStatus status) { // some of these are non-critical-Errors - debug(1, "FlacInputStream: An error occured while decoding. DecoderState is: %s", + debug(1, "FlacInputStream: An error occured while decoding. DecoderState is: %s", FLAC__StreamDecoderErrorStatusString[status]); } @@ -794,7 +794,7 @@ void FlacTrackInfo::play(Audio::Mixer *mixer, Audio::SoundHandle *handle, int st debug(1, "FlacTrackInfo: Audiostream %s could not seek to frame %d (ca %d secs)", _file->name(), startFrame, startFrame/75); flac->finish(); } - delete flac; + delete flac; } FlacTrackInfo::~FlacTrackInfo() diff --git a/sound/fmopl.cpp b/sound/fmopl.cpp index ddba9dbfef..056459ceea 100644 --- a/sound/fmopl.cpp +++ b/sound/fmopl.cpp @@ -451,7 +451,7 @@ inline void OPL_CALC_CH(OPL_CH *CH) { env_out=OPL_CALC_SLOT(SLOT); if(env_out < (uint)(EG_ENT - 1)) { /* PG */ - if(SLOT->vib) + if(SLOT->vib) SLOT->Cnt += (SLOT->Incr * vib / VIB_RATE); else SLOT->Cnt += SLOT->Incr; @@ -487,7 +487,7 @@ inline void OPL_CALC_CH(OPL_CH *CH) { inline void OPL_CALC_RH(OPL_CH *CH) { uint env_tam, env_sd, env_top, env_hh; int whitenoise = int(oplRnd.getRandomNumber(1) * (WHITE_NOISE_db / EG_STEP)); - + int tone8; OPL_SLOT *SLOT; @@ -584,7 +584,7 @@ static void init_timetables(FM_OPL *OPL, int ARRATE, int DRRATE) { OPL->AR_TABLE[i] = OPL->DR_TABLE[i] = 0; for (i = 4; i <= 60; i++){ rate = OPL->freqbase; /* frequency rate */ - if(i < 60) + if(i < 60) rate *= 1.0 + (i & 3) * 0.25; /* b0-1 : x1 , x1.25 , x1.5 , x1.75 */ rate *= 1 << ((i >> 2) - 1); /* b2-5 : shift bit */ rate *= (double)(EG_ENT << ENV_BITS); @@ -973,7 +973,7 @@ void YM3812UpdateOne(FM_OPL *OPL, int16 *buffer, int length) { ARM_CALL(ARM_COMMON, PNO_DATA()) ARM_END(); #endif - + int i; int data; int16 *buf = buffer; diff --git a/sound/fmopl.h b/sound/fmopl.h index c65fd553ce..b80a4eb2d0 100644 --- a/sound/fmopl.h +++ b/sound/fmopl.h @@ -59,7 +59,7 @@ typedef struct fm_opl_slot { uint mul; /* multiple :ML_TABLE[ML] */ uint Cnt; /* frequency count */ uint Incr; /* frequency step */ - + /* envelope generator state */ uint8 eg_typ;/* envelope type flag */ uint8 evm; /* envelope phase */ @@ -116,7 +116,7 @@ typedef struct fm_opl_f { /* Rythm sention */ uint8 rythm; /* Rythm mode , key flag */ - + /* time tables */ int AR_TABLE[75]; /* atttack rate tables */ int DR_TABLE[75]; /* decay rate tables */ diff --git a/sound/mididrv.cpp b/sound/mididrv.cpp index 45a7031f7a..fe24f51a1e 100644 --- a/sound/mididrv.cpp +++ b/sound/mididrv.cpp @@ -136,7 +136,7 @@ int MidiDriver::detectMusicDriver(int midiFlags) { musicDriver = MD_ETUDE; #elif defined(_WIN32_WCE) || defined(UNIX) || defined(X11_BACKEND) || defined (__SYMBIAN32__) // Always use MIDI emulation via adlib driver on CE and UNIX device - + // TODO: We should, for the Unix targets, attempt to detect // whether a sequencer is available, and use it instead. musicDriver = MD_ADLIB; @@ -195,7 +195,7 @@ MidiDriver *MidiDriver::createMidi(int midiDriver) { case MD_ZODIAC: return MidiDriver_Zodiac_create(); #endif #endif -#if defined(WIN32) && !defined(_WIN32_WCE) && !defined(__SYMBIAN32__) +#if defined(WIN32) && !defined(_WIN32_WCE) && !defined(__SYMBIAN32__) case MD_WINDOWS: return MidiDriver_WIN_create(); #endif #if defined(__MORPHOS__) diff --git a/sound/mididrv.h b/sound/mididrv.h index 07b6e18611..556c0c2414 100644 --- a/sound/mididrv.h +++ b/sound/mididrv.h @@ -149,7 +149,7 @@ public: // Timing functions - MidiDriver now operates timers virtual void setTimerCallback(void *timer_param, Common::Timer::TimerProc timer_proc) = 0; - + /** The time in microseconds between invocations of the timer callback. */ virtual uint32 getBaseTempo(void) = 0; diff --git a/sound/midiparser_xmidi.cpp b/sound/midiparser_xmidi.cpp index 8d8c2e9ea1..3ee08ccb7d 100644 --- a/sound/midiparser_xmidi.cpp +++ b/sound/midiparser_xmidi.cpp @@ -26,7 +26,7 @@ /** * The XMIDI version of MidiParser. - * + * * Much of this code is adapted from the XMIDI implementation from the exult * project. */ @@ -52,7 +52,7 @@ public: uint32 MidiParser_XMIDI::readVLQ2(byte * &pos) { uint32 value = 0; int i; - + for (i = 0; i < 4; ++i) { if (pos[0] & 0x80) break; @@ -140,12 +140,12 @@ bool MidiParser_XMIDI::loadMusic(byte *data, uint32 size) { if (!memcmp(pos, "FORM", 4)) { pos += 4; - // Read length of + // Read length of len = read4high(pos); start = pos; // XDIRless XMIDI, we can handle them here. - if (!memcmp(pos, "XMID", 4)) { + if (!memcmp(pos, "XMID", 4)) { warning("XMIDI doesn't have XDIR"); pos += 4; _num_tracks = 1; diff --git a/sound/mixer.cpp b/sound/mixer.cpp index f97c563ccf..c32675a85d 100644 --- a/sound/mixer.cpp +++ b/sound/mixer.cpp @@ -114,7 +114,7 @@ Mixer::Mixer() { _volumeForSoundType[i] = kMaxMixerVolume; _paused = false; - + for (i = 0; i != NUM_CHANNELS; i++) _channels[i] = 0; @@ -144,7 +144,7 @@ void Mixer::setupPremix(AudioStream *stream, SoundType type) { delete _premixChannel; _premixChannel = 0; - + if (stream == 0) return; @@ -399,7 +399,7 @@ void Mixer::setVolumeForSoundType(SoundType type, int volume) { volume = kMaxMixerVolume; else if (volume < 0) volume = 0; - + // TODO: Maybe we should do logarithmic (not linear) volume // scaling? See also Player_V2::setMasterVolume @@ -408,7 +408,7 @@ void Mixer::setVolumeForSoundType(SoundType type, int volume) { int Mixer::getVolumeForSoundType(SoundType type) const { assert(0 <= type && type < ARRAYSIZE(_volumeForSoundType)); - + return _volumeForSoundType[type]; } @@ -462,7 +462,7 @@ void Channel::mix(int16 *data, uint len) { // balance value ranges from -127 to 127. The mixer (music/sound) // volume is in the range 0 - kMaxMixerVolume. // Hence, the vol_l/vol_r values will be in that range, too - + int vol = _mixer->getVolumeForSoundType(_type) * _volume; st_volume_t vol_l, vol_r; diff --git a/sound/mixer.h b/sound/mixer.h index 001753301a..72f881d626 100644 --- a/sound/mixer.h +++ b/sound/mixer.h @@ -69,7 +69,7 @@ public: /** loop the audio */ FLAG_LOOP = 1 << 6 }; - + enum SoundType { kPlainSoundType = 0, @@ -77,7 +77,7 @@ public: kSFXSoundType = 2, kSpeechSoundType = 3 }; - + enum { kMaxChannelVolume = 255, kMaxMixerVolume = 256 @@ -98,7 +98,7 @@ private: int _volumeForSoundType[4]; bool _paused; - + uint32 _handleSeed; Channel *_channels[NUM_CHANNELS]; diff --git a/sound/mp3.cpp b/sound/mp3.cpp index 792f8f6bf4..4c9f238eed 100644 --- a/sound/mp3.cpp +++ b/sound/mp3.cpp @@ -64,7 +64,7 @@ public: bool endOfData() const { return eosIntern(); } bool isStereo() const { return _isStereo; } - + int getRate() const { return _frame.header.samplerate; } #ifdef __SYMBIAN32__ // Used to store the last position stream was read for symbian @@ -193,7 +193,7 @@ bool MP3InputStream::init() { warning("MP3InputStream: Cannot determine number of channels"); return false; } - + return true; } @@ -256,7 +256,7 @@ void MP3InputStream::refill(bool first) { mad_timer_t frame_duration = _frame.header.duration; mad_timer_negate(&frame_duration); mad_timer_add(&_duration, frame_duration); - + if (!first && mad_timer_compare(_duration, mad_timer_zero) <= 0) _size = -1; // Mark for EOF } diff --git a/sound/mpu401.cpp b/sound/mpu401.cpp index 460aed5f52..c46c3d60ac 100644 --- a/sound/mpu401.cpp +++ b/sound/mpu401.cpp @@ -90,7 +90,7 @@ MidiDriver_MPU401::MidiDriver_MPU401() : _timer_proc (0), _channel_mask (0xFFFF) // Permit all 16 channels by default { - + uint i; for (i = 0; i < ARRAYSIZE(_midi_channels); ++i) { _midi_channels [i].init (this, i); diff --git a/sound/rate.cpp b/sound/rate.cpp index eedc283cc6..997aa7e7cf 100644 --- a/sound/rate.cpp +++ b/sound/rate.cpp @@ -209,10 +209,10 @@ public: 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; st_size_t len; - + if (stereo) osamp *= 2; @@ -225,7 +225,7 @@ public: // Read up to 'osamp' samples into our temporary buffer len = input.readBuffer(_buffer, osamp); - + // Mix the data into the output buffer ptr = _buffer; while (len--) { @@ -241,7 +241,7 @@ public: // output left channel clampedAdd(*obuf++, (tmp0 * (int)vol_l) / Audio::Mixer::kMaxMixerVolume); - + // output right channel clampedAdd(*obuf++, (tmp1 * (int)vol_r) / Audio::Mixer::kMaxMixerVolume); } diff --git a/sound/softsynth/adlib.cpp b/sound/softsynth/adlib.cpp index 5f0d0435bd..8fb7aea19e 100644 --- a/sound/softsynth/adlib.cpp +++ b/sound/softsynth/adlib.cpp @@ -195,7 +195,7 @@ struct AdlibVoice { Struct11 _s11a; Struct10 _s10b; Struct11 _s11b; - + AdlibVoice() { memset(this, 0, sizeof(AdlibVoice)); } }; @@ -461,14 +461,14 @@ static byte gm_percussion_to_fm[39][30] = { }; static const byte gm_percussion_lookup[128] = { - 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, - 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, - 0xFF, 0xFF, 0xFF, 0x00, 0x01, 0x02, 0x03, 0x04, 0x05, 0x06, 0x07, 0x08, 0x09, 0x0A, 0x0B, 0x0C, - 0x0D, 0x0E, 0x0F, 0x10, 0x11, 0x12, 0x13, 0x14, 0x15, 0x16, 0xFF, 0xFF, 0x17, 0x18, 0x19, 0x1A, - 0x1B, 0x1C, 0x1D, 0x1E, 0x1F, 0x20, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0x21, 0x22, 0x23, 0xFF, 0xFF, - 0x24, 0x25, 0xFF, 0xFF, 0xFF, 0x26, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, - 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, - 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, + 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, + 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, + 0xFF, 0xFF, 0xFF, 0x00, 0x01, 0x02, 0x03, 0x04, 0x05, 0x06, 0x07, 0x08, 0x09, 0x0A, 0x0B, 0x0C, + 0x0D, 0x0E, 0x0F, 0x10, 0x11, 0x12, 0x13, 0x14, 0x15, 0x16, 0xFF, 0xFF, 0x17, 0x18, 0x19, 0x1A, + 0x1B, 0x1C, 0x1D, 0x1E, 0x1F, 0x20, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0x21, 0x22, 0x23, 0xFF, 0xFF, + 0x24, 0x25, 0xFF, 0xFF, 0xFF, 0x26, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, + 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, + 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, }; static byte lookup_table[64][32]; @@ -551,7 +551,7 @@ public: void send(byte channel, uint32 b); // Supports higher than channel 15 uint32 property(int prop, uint32 param); - void setPitchBendRange(byte channel, uint range); + void setPitchBendRange(byte channel, uint range); void sysEx_customInstrument(byte channel, uint32 type, byte *instr); MidiChannel *allocateChannel(); @@ -798,7 +798,7 @@ void AdlibPercussionChannel::noteOn(byte note, byte velocity) { // MidiDriver method implementations -MidiDriver_ADLIB::MidiDriver_ADLIB(Audio::Mixer *mixer) +MidiDriver_ADLIB::MidiDriver_ADLIB(Audio::Mixer *mixer) : MidiDriver_Emulated(mixer) { uint i; @@ -865,7 +865,7 @@ void MidiDriver_ADLIB::close() { // Turn off the OPL emulation // YM3812Shutdown(); - + free(_adlib_reg_cache); } @@ -1308,7 +1308,7 @@ AdlibVoice *MidiDriver_ADLIB::allocate_voice(byte pri) { } /* V3 games don't have note priorities, first comes wins. */ - if (_game_SmallHeader) + if (_game_SmallHeader) return NULL; if (best) diff --git a/sound/softsynth/emumidi.h b/sound/softsynth/emumidi.h index 579aebeeff..75559530f3 100644 --- a/sound/softsynth/emumidi.h +++ b/sound/softsynth/emumidi.h @@ -46,10 +46,10 @@ protected: public: MidiDriver_Emulated(Audio::Mixer *mixer) : _mixer(mixer) { _isOpen = false; - + _timerProc = 0; _timerParam = 0; - + _nextTick = 0; _samplesPerTick = 0; @@ -89,7 +89,7 @@ public: step = (_nextTick >> FIXP_SHIFT); generateSamples(data, step); - + _nextTick -= step << FIXP_SHIFT; if (!(_nextTick >> FIXP_SHIFT)) { if (_timerProc) diff --git a/sound/softsynth/fluidsynth.cpp b/sound/softsynth/fluidsynth.cpp index 5bcf0993af..563dbd84f1 100644 --- a/sound/softsynth/fluidsynth.cpp +++ b/sound/softsynth/fluidsynth.cpp @@ -73,7 +73,7 @@ MidiDriver_FluidSynth::MidiDriver_FluidSynth(Audio::Mixer *mixer) } // It ought to be possible to get FluidSynth to generate samples at - // lower + // lower _outputRate = _mixer->getOutputRate(); if (_outputRate < 22050) diff --git a/sound/softsynth/mt32.cpp b/sound/softsynth/mt32.cpp index 94373b980a..2cef1cbedf 100644 --- a/sound/softsynth/mt32.cpp +++ b/sound/softsynth/mt32.cpp @@ -238,7 +238,7 @@ int MidiDriver_MT32::open() { return MERR_ALREADY_OPEN; MidiDriver_Emulated::open(); - + memset(&prop, 0, sizeof(prop)); prop.sampleRate = getRate(); prop.useReverb = true; @@ -253,10 +253,10 @@ int MidiDriver_MT32::open() { _synth = new MT32Emu::Synth(); _initialising = true; const byte dummy_palette[] = { - 0, 0, 0, 0, - 0, 0, 171, 0, - 0, 171, 0, 0, - 0, 171, 171, 0, + 0, 0, 0, 0, + 0, 0, 171, 0, + 0, 171, 0, 0, + 0, 171, 171, 0, 171, 0, 0, 0 }; @@ -470,7 +470,7 @@ void MidiDriver_ThreadedMT32::onTimer() { MidiDriver *MidiDriver_MT32_create(Audio::Mixer *mixer) { // HACK: It will stay here until engine plugin loader overhaul - if (ConfMan.hasKey("extrapath")) + if (ConfMan.hasKey("extrapath")) Common::File::addDefaultDirectory(ConfMan.get("extrapath")); return new MidiDriver_MT32(mixer); } diff --git a/sound/softsynth/mt32/freeverb.cpp b/sound/softsynth/mt32/freeverb.cpp index 42865c1e71..0a75e7bc70 100644 --- a/sound/softsynth/mt32/freeverb.cpp +++ b/sound/softsynth/mt32/freeverb.cpp @@ -26,7 +26,7 @@ // Comb filter implementation // -// Written by +// Written by // http://www.dreampoint.co.uk // This code is public domain @@ -39,7 +39,7 @@ comb::comb() { } void comb::setbuffer(float *buf, int size) { - buffer = buf; + buffer = buf; bufsize = size; } @@ -49,7 +49,7 @@ void comb::mute() { } void comb::setdamp(float val) { - damp1 = val; + damp1 = val; damp2 = 1 - val; } @@ -72,7 +72,7 @@ allpass::allpass() { } void allpass::setbuffer(float *buf, int size) { - buffer = buf; + buffer = buf; bufsize = size; } diff --git a/sound/softsynth/mt32/freeverb.h b/sound/softsynth/mt32/freeverb.h index 2a5d662a44..b34413f188 100644 --- a/sound/softsynth/mt32/freeverb.h +++ b/sound/softsynth/mt32/freeverb.h @@ -101,10 +101,10 @@ private: inline float allpass::process(float input) { float output; float bufout; - + bufout = buffer[bufidx]; undenormalise(bufout); - + output = -input + bufout; buffer[bufidx] = input + (bufout * feedback); @@ -196,7 +196,7 @@ private: float width; float mode; - // The following are all declared inline + // The following are all declared inline // to remove the need for dynamic allocation // with its subsequent error-checking messiness diff --git a/sound/softsynth/mt32/part.h b/sound/softsynth/mt32/part.h index 8a5612e32d..54c4999653 100644 --- a/sound/softsynth/mt32/part.h +++ b/sound/softsynth/mt32/part.h @@ -42,7 +42,7 @@ private: PatchCache patchCache[4]; - float bend; // -1.0 .. +1.0 + float bend; // -1.0 .. +1.0 dpoly polyTable[MT32EMU_MAX_POLY]; diff --git a/sound/softsynth/mt32/synth.cpp b/sound/softsynth/mt32/synth.cpp index 66832b0e59..62356dcda9 100644 --- a/sound/softsynth/mt32/synth.cpp +++ b/sound/softsynth/mt32/synth.cpp @@ -824,31 +824,31 @@ void Synth::readMemoryRegion(const MemoryRegion *region, Bit32u addr, Bit32u len switch(region->type) { case MR_PatchTemp: - for (m = 0; m < len; m++) + for (m = 0; m < len; m++) data[m] = ((Bit8u *)&mt32ram.patchSettings[first])[off + m]; break; case MR_RhythmTemp: - for (m = 0; m < len; m++) + for (m = 0; m < len; m++) data[m] = ((Bit8u *)&mt32ram.rhythmSettings[first])[off + m]; break; case MR_TimbreTemp: - for (m = 0; m < len; m++) + for (m = 0; m < len; m++) data[m] = ((Bit8u *)&mt32ram.timbreSettings[first])[off + m]; break; case MR_Patches: - for (m = 0; m < len; m++) + for (m = 0; m < len; m++) data[m] = ((Bit8u *)&mt32ram.patches[first])[off + m]; break; case MR_Timbres: - for (m = 0; m < len; m++) + for (m = 0; m < len; m++) data[m] = ((Bit8u *)&mt32ram.timbres[first])[off + m]; break; case MR_System: - for (m = 0; m < len; m++) + for (m = 0; m < len; m++) data[m] = ((Bit8u *)&mt32ram.system)[m + off]; break; default: - for (m = 0; m < len; m += 2) { + for (m = 0; m < len; m += 2) { data[m] = 0xff; if (m + 1 < len) { data[m+1] = (Bit8u)region->type; diff --git a/sound/softsynth/mt32/tables.cpp b/sound/softsynth/mt32/tables.cpp index e12f306cf9..006f91d0b8 100644 --- a/sound/softsynth/mt32/tables.cpp +++ b/sound/softsynth/mt32/tables.cpp @@ -170,7 +170,7 @@ void Tables::initEnvelopes(float samplerate) { } envDeltaMaxTime[lf] = (int)cap; } - + // This (approximately) represents the time durations when the target level is 0. // Not sure why this is a special case, but it's seen to be from the real thing. diff --git a/sound/softsynth/ym2612.cpp b/sound/softsynth/ym2612.cpp index e37b50170e..80c0cad8e8 100644 --- a/sound/softsynth/ym2612.cpp +++ b/sound/softsynth/ym2612.cpp @@ -106,7 +106,7 @@ protected: int _frequencyOffs; int _frequency; int _algorithm; - + int *_buffer; int _buflen; diff --git a/sound/vorbis.cpp b/sound/vorbis.cpp index 0f771cb5e3..7c18a83a8e 100644 --- a/sound/vorbis.cpp +++ b/sound/vorbis.cpp @@ -126,7 +126,7 @@ static ov_callbacks g_File_wrap = { VorbisTrackInfo::VorbisTrackInfo(File *file) { - + _file = file; if (openTrack()) { warning("Invalid file format"); @@ -149,9 +149,9 @@ bool VorbisTrackInfo::openTrack() { f->len = _file->size(); f->curr_pos = 0; _file->seek(0); - + bool err = (ov_open_callbacks((void *) f, &_ov_file, NULL, 0, g_File_wrap) < 0); - + if (err) { delete f; } else { @@ -218,7 +218,7 @@ class VorbisInputStream : public AudioStream { const int16 *_bufferEnd; const int16 *_pos; bool _deleteFileAfterUse; - + void refill(); inline bool eosIntern() const; public: @@ -229,7 +229,7 @@ public: bool endOfData() const { return eosIntern(); } bool isStereo() const { return _numChannels >= 2; } - + int getRate() const { return ov_info(_ov_file, -1)->rate; } }; @@ -240,7 +240,7 @@ public: #endif -VorbisInputStream::VorbisInputStream(OggVorbis_File *file, int duration, bool deleteFileAfterUse) +VorbisInputStream::VorbisInputStream(OggVorbis_File *file, int duration, bool deleteFileAfterUse) : _ov_file(file), _bufferEnd(_buffer + ARRAYSIZE(_buffer)), _deleteFileAfterUse(deleteFileAfterUse) { diff --git a/sound/wave.cpp b/sound/wave.cpp index 42c1bd34db..97dd030ab8 100644 --- a/sound/wave.cpp +++ b/sound/wave.cpp @@ -53,7 +53,7 @@ bool loadWAVFromStream(Common::SeekableReadStream &stream, int &size, int &rate, warning("getWavInfo: No 'fmt' header"); return false; } - + uint32 fmtLength = stream.readUint32LE(); if (fmtLength < 16) { // A valid fmt chunk always contains at least 16 bytes @@ -79,7 +79,7 @@ bool loadWAVFromStream(Common::SeekableReadStream &stream, int &size, int &rate, if (wavType != 0) *wavType = type; -#if 0 +#if 0 printf("WAVE information:\n"); printf(" total size: %d\n", wavLength); printf(" fmt size: %d\n", fmtLength); @@ -118,7 +118,7 @@ bool loadWAVFromStream(Common::SeekableReadStream &stream, int &size, int &rate, warning("getWavInfo: unsupported bitsPerSample %d", bitsPerSample); return false; } - + if (numChannels == 2) flags |= Audio::Mixer::FLAG_STEREO; else if (numChannels != 1) { @@ -145,7 +145,7 @@ bool loadWAVFromStream(Common::SeekableReadStream &stream, int &size, int &rate, printf(" found a '%s' tag of size %d\n", buf, offset); #endif } while (memcmp(buf, "data", 4) != 0); - + // Stream now points at 'offset' bytes of sample data... size = offset; @@ -156,10 +156,10 @@ AudioStream *makeWAVStream(Common::SeekableReadStream &stream) { int size, rate; byte flags; uint16 type; - + if (!loadWAVFromStream(stream, size, rate, flags, &type)) return 0; - + if (type == 17) // IMA ADPCM return makeADPCMStream(stream, size, kADPCMIma); -- cgit v1.2.3