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author | Max Horn | 2003-08-05 02:05:37 +0000 |
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committer | Max Horn | 2003-08-05 02:05:37 +0000 |
commit | 1eaf41eda40f1777a5e99c19a027ea025cbaeb4c (patch) | |
tree | 66c67f0f994ee3f64c2fee33269d7d21d6a9ad24 /sound/mixer.cpp | |
parent | 2bc604081e4ef7ac8cf39cdde3a89dd494fbb6e1 (diff) | |
download | scummvm-rg350-1eaf41eda40f1777a5e99c19a027ea025cbaeb4c.tar.gz scummvm-rg350-1eaf41eda40f1777a5e99c19a027ea025cbaeb4c.tar.bz2 scummvm-rg350-1eaf41eda40f1777a5e99c19a027ea025cbaeb4c.zip |
more cleanup (like the previous two, I just moved around code, to clearly separate the old and the new mixer code)
svn-id: r9479
Diffstat (limited to 'sound/mixer.cpp')
-rw-r--r-- | sound/mixer.cpp | 345 |
1 files changed, 190 insertions, 155 deletions
diff --git a/sound/mixer.cpp b/sound/mixer.cpp index fbeef67d6d..1cd63ea5b2 100644 --- a/sound/mixer.cpp +++ b/sound/mixer.cpp @@ -30,31 +30,22 @@ #ifdef NEW_MIXER_CODE #include "rate.h" -#endif class Channel { protected: SoundMixer *_mixer; PlayingSoundHandle *_handle; -#ifdef NEW_MIXER_CODE RateConverter *_converter; AudioInputStream *_input; -#endif public: int _id; Channel(SoundMixer *mixer, PlayingSoundHandle *handle) - : _mixer(mixer), _handle(handle), -#ifdef NEW_MIXER_CODE - _converter(0), _input(0), -#endif - _id(-1) { + : _mixer(mixer), _handle(handle), _converter(0), _input(0), _id(-1) { assert(mixer); } virtual ~Channel() { -#ifdef NEW_MIXER_CODE delete _converter; delete _input; -#endif if (_handle) *_handle = 0; } @@ -63,7 +54,6 @@ public: 10 means that the buffer contains twice 10 sample, each 16 bits, for a total of 40 bytes. */ -#ifdef NEW_MIXER_CODE virtual void mix(int16 *data, uint len) { assert(_input); assert(_converter); @@ -77,9 +67,79 @@ public: const int volume = isMusicChannel() ? _mixer->getMusicVolume() : _mixer->getVolume(); _converter->flow(*_input, data, len, volume); } -#else - virtual void mix(int16 *data, uint len) = 0; + void destroy() { + for (int i = 0; i != SoundMixer::NUM_CHANNELS; i++) + if (_mixer->_channels[i] == this) + _mixer->_channels[i] = 0; + delete this; + } + virtual bool isMusicChannel() const = 0; +}; + +class ChannelRaw : public Channel { + byte *_ptr; +public: + ChannelRaw(SoundMixer *mixer, PlayingSoundHandle *handle, void *sound, uint32 size, uint rate, byte flags, int id, uint32 loopStart, uint32 loopEnd); + ~ChannelRaw(); + bool isMusicChannel() const { return false; } +}; + +class ChannelStream : public Channel { + bool _finished; +public: + ChannelStream(SoundMixer *mixer, PlayingSoundHandle *handle, void *sound, uint32 size, uint rate, byte flags, uint32 buffer_size); + void mix(int16 *data, uint len); + void append(void *sound, uint32 size); + bool isMusicChannel() const { return true; } + void finish() { _finished = true; } +}; + +#ifdef USE_MAD +class ChannelMP3 : public Channel { +public: + ChannelMP3(SoundMixer *mixer, PlayingSoundHandle *handle, File *file, uint size); + bool isMusicChannel() const { return false; } +}; + +class ChannelMP3CDMusic : public Channel { +public: + ChannelMP3CDMusic(SoundMixer *mixer, PlayingSoundHandle *handle, File *file, mad_timer_t duration); + bool isMusicChannel() const { return true; } +}; +#endif // USE_MAD + +#ifdef USE_VORBIS +class ChannelVorbis : public Channel { + bool _is_cd_track; +public: + ChannelVorbis(SoundMixer *mixer, PlayingSoundHandle *handle, OggVorbis_File *ov_file, int duration, bool is_cd_track); + bool isMusicChannel() const { return _is_cd_track; } +}; #endif + + +#else // NEW_MIXER_CODE + +class Channel { +protected: + SoundMixer *_mixer; + PlayingSoundHandle *_handle; +public: + int _id; + Channel(SoundMixer *mixer, PlayingSoundHandle *handle) + : _mixer(mixer), _handle(handle), _id(-1) { + assert(mixer); + } + virtual ~Channel() { + if (_handle) + *_handle = 0; + } + + /* len indicates the number of sample *pairs*. So a value of + 10 means that the buffer contains twice 10 sample, each + 16 bits, for a total of 40 bytes. + */ + virtual void mix(int16 *data, uint len) = 0; void destroy() { for (int i = 0; i != SoundMixer::NUM_CHANNELS; i++) if (_mixer->_channels[i] == this) @@ -91,7 +151,6 @@ public: class ChannelRaw : public Channel { byte *_ptr; -#ifndef NEW_MIXER_CODE byte _flags; uint32 _pos; uint32 _size; @@ -100,20 +159,16 @@ class ChannelRaw : public Channel { uint32 _realSize, _rate; byte *_loop_ptr; uint32 _loop_size; -#endif public: ChannelRaw(SoundMixer *mixer, PlayingSoundHandle *handle, void *sound, uint32 size, uint rate, byte flags, int id, uint32 loopStart, uint32 loopEnd); ~ChannelRaw(); -#ifndef NEW_MIXER_CODE void mix(int16 *data, uint len); -#endif bool isMusicChannel() const { return false; } }; class ChannelStream : public Channel { -#ifndef NEW_MIXER_CODE byte *_ptr; byte *_endOfData; byte *_endOfBuffer; @@ -123,7 +178,6 @@ class ChannelStream : public Channel { uint32 _bufferSize; uint32 _rate; byte _flags; -#endif bool _finished; public: @@ -136,24 +190,7 @@ public: void finish() { _finished = true; } }; - #ifdef USE_MAD - -#ifdef NEW_MIXER_CODE - -class ChannelMP3 : public Channel { -public: - ChannelMP3(SoundMixer *mixer, PlayingSoundHandle *handle, File *file, uint size); - bool isMusicChannel() const { return false; } -}; - -class ChannelMP3CDMusic : public Channel { -public: - ChannelMP3CDMusic(SoundMixer *mixer, PlayingSoundHandle *handle, File *file, mad_timer_t duration); - bool isMusicChannel() const { return true; } -}; - -#else // NEW_MIXER_CODE class ChannelMP3Common : public Channel { protected: byte *_ptr; @@ -190,29 +227,22 @@ public: void mix(int16 *data, uint len); bool isMusicChannel() const { return true; } }; -#endif // NEW_MIXER_CODE - #endif // USE_MAD - #ifdef USE_VORBIS class ChannelVorbis : public Channel { -#ifndef NEW_MIXER_CODE OggVorbis_File *_ov_file; int _end_pos; -#endif bool _is_cd_track; - public: ChannelVorbis(SoundMixer *mixer, PlayingSoundHandle *handle, OggVorbis_File *ov_file, int duration, bool is_cd_track); - -#ifndef NEW_MIXER_CODE void mix(int16 *data, uint len); -#endif bool isMusicChannel() const { return _is_cd_track; } }; #endif +#endif + SoundMixer::SoundMixer() { _syst = 0; @@ -460,8 +490,116 @@ void SoundMixer::setMusicVolume(int volume) { } #ifdef NEW_MIXER_CODE -#define clamped_add_16(a, b) clampedAdd(a, b) -#else + +/* RAW mixer */ +ChannelRaw::ChannelRaw(SoundMixer *mixer, PlayingSoundHandle *handle, void *sound, uint32 size, uint rate, byte flags, int id, uint32 loopStart, uint32 loopEnd) + : Channel(mixer, handle) { + _id = id; + _ptr = (byte *)sound; + + // Create the input stream + if (flags & SoundMixer::FLAG_LOOP) { + if (loopEnd == 0) { + _input = makeLinearInputStream(flags, _ptr, size, 0, size); + } else { + assert(loopStart < loopEnd && loopEnd <= size); + _input = makeLinearInputStream(flags, _ptr, size, loopStart, loopEnd - loopStart); + } + } else { + _input = makeLinearInputStream(flags, _ptr, size, 0, 0); + } + + // Get a rate converter instance + _converter = makeRateConverter(rate, mixer->getOutputRate(), _input->isStereo(), (flags & SoundMixer::FLAG_REVERSE_STEREO) != 0); + + if (!(flags & SoundMixer::FLAG_AUTOFREE)) + _ptr = 0; +} + +ChannelRaw::~ChannelRaw() { + free(_ptr); +} + +ChannelStream::ChannelStream(SoundMixer *mixer, PlayingSoundHandle *handle, + void *sound, uint32 size, uint rate, + byte flags, uint32 buffer_size) + : Channel(mixer, handle) { + assert(size <= buffer_size); + + // Create the input stream + _input = makeWrappedInputStream(flags, buffer_size); + + // Append the initial data + ((WrappedAudioInputStream *)_input)->append((const byte *)sound, size); + + // Get a rate converter instance + _converter = makeRateConverter(rate, mixer->getOutputRate(), _input->isStereo(), (flags & SoundMixer::FLAG_REVERSE_STEREO) != 0); + + _finished = false; +} + +void ChannelStream::append(void *data, uint32 len) { + ((WrappedAudioInputStream *)_input)->append((const byte *)data, len); +} + +void ChannelStream::mix(int16 *data, uint len) { + assert(_input); + assert(_converter); + + if (_input->eos()) { + // TODO: call drain method + + // Normally, the stream stays around even if all its data is used up. + // This is in case more data is streamed into it. To make the stream + // go away, one can either stop() it (which takes effect immediately, + // ignoring any remaining sound data), or finish() it, which means + // it will finish playing before it terminates itself. + if (_finished) { + destroy(); + } + + return; + } + + const int volume = _mixer->getVolume(); // FIXME: Shouldn't this be music volume instead?? +// const int volume = isMusicChannel() ? _mixer->getMusicVolume() : _mixer->getVolume(); + _converter->flow(*_input, data, len, volume); +} + +#ifdef USE_MAD +ChannelMP3::ChannelMP3(SoundMixer *mixer, PlayingSoundHandle *handle, File *file, uint size) + : Channel(mixer, handle) { + // Create the input stream + _input = makeMP3Stream(file, mad_timer_zero, size); + + // Get a rate converter instance + _converter = makeRateConverter(_input->getRate(), mixer->getOutputRate(), _input->isStereo()); +} + +ChannelMP3CDMusic::ChannelMP3CDMusic(SoundMixer *mixer, PlayingSoundHandle *handle, File *file, mad_timer_t duration) + : Channel(mixer, handle) { + // Create the input stream + _input = makeMP3Stream(file, duration, 0); + + // Get a rate converter instance + _converter = makeRateConverter(_input->getRate(), mixer->getOutputRate(), _input->isStereo()); +} +#endif // USE_MAD + +#ifdef USE_VORBIS +ChannelVorbis::ChannelVorbis(SoundMixer *mixer, PlayingSoundHandle *handle, OggVorbis_File *ov_file, int duration, bool is_cd_track) + : Channel(mixer, handle) { + // Create the input stream + _input = makeVorbisStream(ov_file, duration); + + // Get a rate converter instance + _converter = makeRateConverter(_input->getRate(), mixer->getOutputRate(), _input->isStereo()); + _is_cd_track = is_cd_track; +} +#endif // USE_VORBIS + +#else // NEW_MIXER_CODE + /* * Class that performs cubic interpolation on integer data. * It is expected that the data is equidistant, i.e. all have the same @@ -674,7 +812,6 @@ static int16 mixer_element_size[] = { 2, 2, 4, 4 }; -#endif /* RAW mixer */ ChannelRaw::ChannelRaw(SoundMixer *mixer, PlayingSoundHandle *handle, void *sound, uint32 size, uint rate, byte flags, int id, uint32 loopStart, uint32 loopEnd) @@ -682,25 +819,6 @@ ChannelRaw::ChannelRaw(SoundMixer *mixer, PlayingSoundHandle *handle, void *soun _id = id; _ptr = (byte *)sound; -#ifdef NEW_MIXER_CODE - // Create the input stream - if (flags & SoundMixer::FLAG_LOOP) { - if (loopEnd == 0) { - _input = makeLinearInputStream(flags, _ptr, size, 0, size); - } else { - assert(loopStart < loopEnd && loopEnd <= size); - _input = makeLinearInputStream(flags, _ptr, size, loopStart, loopEnd - loopStart); - } - } else { - _input = makeLinearInputStream(flags, _ptr, size, 0, 0); - } - - // Get a rate converter instance - _converter = makeRateConverter(rate, mixer->getOutputRate(), _input->isStereo(), (flags & SoundMixer::FLAG_REVERSE_STEREO) != 0); - - if (!(flags & SoundMixer::FLAG_AUTOFREE)) - _ptr = 0; -#else _flags = flags; _pos = 0; _fpPos = 0; @@ -722,19 +840,13 @@ ChannelRaw::ChannelRaw(SoundMixer *mixer, PlayingSoundHandle *handle, void *soun _loop_ptr = _ptr; _loop_size = _size; } -#endif } ChannelRaw::~ChannelRaw() { -#ifdef NEW_MIXER_CODE - free(_ptr); -#else if (_flags & SoundMixer::FLAG_AUTOFREE) free(_ptr); -#endif } -#ifndef NEW_MIXER_CODE void ChannelRaw::mix(int16 *data, uint len) { byte *s, *end; @@ -760,26 +872,15 @@ void ChannelRaw::mix(int16 *data, uint len) { } } } -#endif #define WARP_WORKAROUND 50000 -ChannelStream::ChannelStream(SoundMixer *mixer, PlayingSoundHandle *handle, void *sound, uint32 size, uint rate, - byte flags, uint32 buffer_size) +ChannelStream::ChannelStream(SoundMixer *mixer, PlayingSoundHandle *handle, + void *sound, uint32 size, uint rate, + byte flags, uint32 buffer_size) : Channel(mixer, handle) { assert(size <= buffer_size); -#ifdef NEW_MIXER_CODE - - // Create the input stream - _input = makeWrappedInputStream(flags, buffer_size); - - // Append the initial data - ((WrappedAudioInputStream *)_input)->append((const byte *)sound, size); - - // Get a rate converter instance - _converter = makeRateConverter(rate, mixer->getOutputRate(), _input->isStereo(), (flags & SoundMixer::FLAG_REVERSE_STEREO) != 0); -#else _flags = flags; _bufferSize = buffer_size; _ptr = (byte *)malloc(_bufferSize + WARP_WORKAROUND); @@ -795,20 +896,14 @@ ChannelStream::ChannelStream(SoundMixer *mixer, PlayingSoundHandle *handle, void size >>= 1, rate = (rate >> 1) + 1; _rate = rate; -#endif _finished = false; } ChannelStream::~ChannelStream() { -#ifndef NEW_MIXER_CODE free(_ptr); -#endif } void ChannelStream::append(void *data, uint32 len) { -#ifdef NEW_MIXER_CODE - ((WrappedAudioInputStream *)_input)->append((const byte *)data, len); -#else if (_endOfData + len > _endOfBuffer) { /* Wrap-around case */ uint32 size_to_end_of_buffer = _endOfBuffer - _endOfData; @@ -828,33 +923,9 @@ void ChannelStream::append(void *data, uint32 len) { memcpy(_endOfData, data, len); _endOfData += len; } -#endif } void ChannelStream::mix(int16 *data, uint len) { -#ifdef NEW_MIXER_CODE - assert(_input); - assert(_converter); - - if (_input->eos()) { - // TODO: call drain method - - // Normally, the stream stays around even if all its data is used up. - // This is in case more data is streamed into it. To make the stream - // go away, one can either stop() it (which takes effect immediately, - // ignoring any remaining sound data), or finish() it, which means - // it will finish playing before it terminates itself. - if (_finished) { - destroy(); - } - - return; - } - - const int volume = _mixer->getVolume(); // FIXME: Shouldn't this be music volume instead?? -// const int volume = isMusicChannel() ? _mixer->getMusicVolume() : _mixer->getVolume(); - _converter->flow(*_input, data, len, volume); -#else if (_pos == _endOfData) { // Normally, the stream stays around even if all its data is used up. // This is in case more data is streamed into it. To make the stream @@ -902,32 +973,9 @@ void ChannelStream::mix(int16 *data, uint len) { mixProc(data, len, _pos, _fpPos, _fpSpeed, _mixer->getVolume(), _endOfData, (_flags & SoundMixer::FLAG_REVERSE_STEREO) ? true : false); } } -#endif } #ifdef USE_MAD - -#ifdef NEW_MIXER_CODE -ChannelMP3::ChannelMP3(SoundMixer *mixer, PlayingSoundHandle *handle, File *file, uint size) - : Channel(mixer, handle) { - // Create the input stream - _input = makeMP3Stream(file, mad_timer_zero, size); - - // Get a rate converter instance - _converter = makeRateConverter(_input->getRate(), mixer->getOutputRate(), _input->isStereo()); -} - -ChannelMP3CDMusic::ChannelMP3CDMusic(SoundMixer *mixer, PlayingSoundHandle *handle, File *file, mad_timer_t duration) - : Channel(mixer, handle) { - // Create the input stream - _input = makeMP3Stream(file, duration, 0); - - // Get a rate converter instance - _converter = makeRateConverter(_input->getRate(), mixer->getOutputRate(), _input->isStereo()); -} - -#else // NEW_MIXER_CODE - ChannelMP3Common::ChannelMP3Common(SoundMixer *mixer, PlayingSoundHandle *handle) : Channel(mixer, handle) { mad_stream_init(&_stream); @@ -1137,23 +1185,9 @@ void ChannelMP3CDMusic::mix(int16 *data, uint len) { _posInFrame = 0; } } -#endif // NEW_MIXER_CODE - #endif // USE_MAD #ifdef USE_VORBIS - -#ifdef NEW_MIXER_CODE -ChannelVorbis::ChannelVorbis(SoundMixer *mixer, PlayingSoundHandle *handle, OggVorbis_File *ov_file, int duration, bool is_cd_track) - : Channel(mixer, handle) { - // Create the input stream - _input = makeVorbisStream(ov_file, duration); - - // Get a rate converter instance - _converter = makeRateConverter(_input->getRate(), mixer->getOutputRate(), _input->isStereo()); - _is_cd_track = is_cd_track; -} -#else // NEW_MIXER_CODE ChannelVorbis::ChannelVorbis(SoundMixer *mixer, PlayingSoundHandle *handle, OggVorbis_File *ov_file, int duration, bool is_cd_track) : Channel(mixer, handle) { _ov_file = ov_file; @@ -1222,6 +1256,7 @@ void ChannelVorbis::mix(int16 *data, uint len) { if (eof_flag) destroy(); } +#endif // USE_VORBIS + #endif // NEW_MIXER_CODE -#endif // USE_VORBIS |