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TinySoundFont/tsf.h  view on Meta::CPAN

   THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED,
   INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR
   PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE
   LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,
   TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE
   USE OR OTHER DEALINGS IN THE SOFTWARE.

*/

#ifndef TSF_INCLUDE_TSF_INL
#define TSF_INCLUDE_TSF_INL

#ifdef __cplusplus
extern "C" {
#  define CPP_DEFAULT0 = 0
#else
#  define CPP_DEFAULT0
#endif

//define this if you want the API functions to be static
#ifdef TSF_STATIC
#define TSFDEF static
#else
#define TSFDEF extern
#endif

// The load functions will return a pointer to a struct tsf which all functions
// thereafter take as the first parameter.
// On error the tsf_load* functions will return NULL most likely due to invalid
// data (or if the file did not exist in tsf_load_filename).
typedef struct tsf tsf;

#ifndef TSF_NO_STDIO
// Directly load a SoundFont from a .sf2 file path
TSFDEF tsf* tsf_load_filename(const char* filename);
#endif

// Load a SoundFont from a block of memory
TSFDEF tsf* tsf_load_memory(const void* buffer, int size);

// Stream structure for the generic loading
struct tsf_stream
{
	// Custom data given to the functions as the first parameter
	void* data;

	// Function pointer will be called to read 'size' bytes into ptr (returns number of read bytes)
	int (*read)(void* data, void* ptr, unsigned int size);

	// Function pointer will be called to skip ahead over 'count' bytes (returns 1 on success, 0 on error)
	int (*skip)(void* data, unsigned int count);
};

// Generic SoundFont loading method using the stream structure above
TSFDEF tsf* tsf_load(struct tsf_stream* stream);

// Copy a tsf instance from an existing one, use tsf_close to close it as well.
// All copied tsf instances and their original instance are linked, and share the underlying soundfont.
// This allows loading a soundfont only once, but using it for multiple independent playbacks.
// (This function isn't thread-safe without locking.)
TSFDEF tsf* tsf_copy(tsf* f);

// Free the memory related to this tsf instance
TSFDEF void tsf_close(tsf* f);

// Stop all playing notes immediately and reset all channel parameters
TSFDEF void tsf_reset(tsf* f);

// Returns the preset index from a bank and preset number, or -1 if it does not exist in the loaded SoundFont
TSFDEF int tsf_get_presetindex(const tsf* f, int bank, int preset_number);

// Returns the number of presets in the loaded SoundFont
TSFDEF int tsf_get_presetcount(const tsf* f);

// Returns the name of a preset index >= 0 and < tsf_get_presetcount()
TSFDEF const char* tsf_get_presetname(const tsf* f, int preset_index);

// Returns the name of a preset by bank and preset number
TSFDEF const char* tsf_bank_get_presetname(const tsf* f, int bank, int preset_number);

// Supported output modes by the render methods
enum TSFOutputMode
{
	// Two channels with single left/right samples one after another
	TSF_STEREO_INTERLEAVED,
	// Two channels with all samples for the left channel first then right
	TSF_STEREO_UNWEAVED,
	// A single channel (stereo instruments are mixed into center)
	TSF_MONO
};

// Thread safety:
//
// 1. Rendering / voices:
//
// Your audio output which calls the tsf_render* functions will most likely
// run on a different thread than where the playback tsf_note* functions
// are called. In which case some sort of concurrency control like a
// mutex needs to be used so they are not called at the same time.
// Alternatively, you can pre-allocate a maximum number of voices that can
// play simultaneously by calling tsf_set_max_voices after loading.
// That way memory re-allocation will not happen during tsf_note_on and
// TSF should become mostly thread safe.
// There is a theoretical chance that ending notes would negatively influence
// a voice that is rendering at the time but it is hard to say.
// Also be aware, this has not been tested much.
//
// 2. Channels:
//
// Calls to tsf_channel_set_... functions may allocate new channels
// if no channel with that number was previously used. Make sure to
// create all channels at the beginning as required if you call tsf_render*
// from a different thread.

// Setup the parameters for the voice render methods
//   outputmode: if mono or stereo and how stereo channel data is ordered
//   samplerate: the number of samples per second (output frequency)
//   global_gain_db: volume gain in decibels (>0 means higher, <0 means lower)
TSFDEF void tsf_set_output(tsf* f, enum TSFOutputMode outputmode, int samplerate, float global_gain_db CPP_DEFAULT0);

// Set the global gain as a volume factor
//   global_gain: the desired volume where 1.0 is 100%
TSFDEF void tsf_set_volume(tsf* f, float global_gain);

// Set the maximum number of voices to play simultaneously
// Depending on the soundfond, one note can cause many new voices to be started,
// so don't keep this number too low or otherwise sounds may not play.
//   max_voices: maximum number to pre-allocate and set the limit to
//   (tsf_set_max_voices returns 0 if allocation failed, otherwise 1)
TSFDEF int tsf_set_max_voices(tsf* f, int max_voices);

// Start playing a note
//   preset_index: preset index >= 0 and < tsf_get_presetcount()
//   key: note value between 0 and 127 (60 being middle C)
//   vel: velocity as a float between 0.0 (equal to note off) and 1.0 (full)
//   bank: instrument bank number (alternative to preset_index)
//   preset_number: preset number (alternative to preset_index)
//   (tsf_note_on returns 0 if the allocation of a new voice failed, otherwise 1)
//   (tsf_bank_note_on returns 0 if preset does not exist or allocation failed, otherwise 1)
TSFDEF int tsf_note_on(tsf* f, int preset_index, int key, float vel);
TSFDEF int tsf_bank_note_on(tsf* f, int bank, int preset_number, int key, float vel);

// Stop playing a note
//   (bank_note_off returns 0 if preset does not exist, otherwise 1)
TSFDEF void tsf_note_off(tsf* f, int preset_index, int key);
TSFDEF int  tsf_bank_note_off(tsf* f, int bank, int preset_number, int key);

// Stop playing all notes (end with sustain and release)
TSFDEF void tsf_note_off_all(tsf* f);

// Returns the number of active voices
TSFDEF int tsf_active_voice_count(tsf* f);

// Render output samples into a buffer
// You can either render as signed 16-bit values (tsf_render_short) or
// as 32-bit float values (tsf_render_float)
//   buffer: target buffer of size samples * output_channels * sizeof(type)
//   samples: number of samples to render
//   flag_mixing: if 0 clear the buffer first, otherwise mix into existing data
TSFDEF void tsf_render_short(tsf* f, short* buffer, int samples, int flag_mixing CPP_DEFAULT0);
TSFDEF void tsf_render_float(tsf* f, float* buffer, int samples, int flag_mixing CPP_DEFAULT0);

// Higher level channel based functions, set up channel parameters
//   channel: channel number
//   preset_index: preset index >= 0 and < tsf_get_presetcount()
//   preset_number: preset number (alternative to preset_index)
//   flag_mididrums: 0 for normal channels, otherwise apply MIDI drum channel rules
//   bank: instrument bank number (alternative to preset_index)
//   pan: stereo panning value from 0.0 (left) to 1.0 (right) (default 0.5 center)
//   volume: linear volume scale factor (default 1.0 full)
//   pitch_wheel: pitch wheel position 0 to 16383 (default 8192 unpitched)
//   pitch_range: range of the pitch wheel in semitones (default 2.0, total +/- 2 semitones)
//   tuning: tuning of all playing voices in semitones (default 0.0, standard (A440) tuning)

TinySoundFont/tsf.h  view on Meta::CPAN

	}
	if (e->parameters.keynumToDecay)
	{
		e->parameters.decay += e->parameters.keynumToDecay * (60.0f - midiNoteNumber);
		e->parameters.decay = (e->parameters.decay < -10000.0f ? 0.0f : tsf_timecents2Secsf(e->parameters.decay));
	}
	e->midiVelocity = midiVelocity;
	e->isAmpEnv = isAmpEnv;
	tsf_voice_envelope_nextsegment(e, TSF_SEGMENT_NONE, outSampleRate);
}

static void tsf_voice_envelope_process(struct tsf_voice_envelope* e, int numSamples, float outSampleRate)
{
	if (e->slope)
	{
		if (e->segmentIsExponential) e->level *= TSF_POWF(e->slope, (float)numSamples);
		else e->level += (e->slope * numSamples);
	}
	if ((e->samplesUntilNextSegment -= numSamples) <= 0)
		tsf_voice_envelope_nextsegment(e, e->segment, outSampleRate);
}

static void tsf_voice_lowpass_setup(struct tsf_voice_lowpass* e, float Fc)
{
	// Lowpass filter from http://www.earlevel.com/main/2012/11/26/biquad-c-source-code/
	double K = TSF_TAN(TSF_PI * Fc), KK = K * K;
	double norm = 1 / (1 + K * e->QInv + KK);
	e->a0 = KK * norm;
	e->a1 = 2 * e->a0;
	e->b1 = 2 * (KK - 1) * norm;
	e->b2 = (1 - K * e->QInv + KK) * norm;
}

static float tsf_voice_lowpass_process(struct tsf_voice_lowpass* e, double In)
{
	double Out = In * e->a0 + e->z1; e->z1 = In * e->a1 + e->z2 - e->b1 * Out; e->z2 = In * e->a0 - e->b2 * Out; return (float)Out;
}

static void tsf_voice_lfo_setup(struct tsf_voice_lfo* e, float delay, int freqCents, float outSampleRate)
{
	e->samplesUntil = (int)(delay * outSampleRate);
	e->delta = (4.0f * tsf_cents2Hertz((float)freqCents) / outSampleRate);
	e->level = 0;
}

static void tsf_voice_lfo_process(struct tsf_voice_lfo* e, int blockSamples)
{
	if (e->samplesUntil > blockSamples) { e->samplesUntil -= blockSamples; return; }
	e->level += e->delta * blockSamples;
	if      (e->level >  1.0f) { e->delta = -e->delta; e->level =  2.0f - e->level; }
	else if (e->level < -1.0f) { e->delta = -e->delta; e->level = -2.0f - e->level; }
}

static void tsf_voice_kill(struct tsf_voice* v)
{
	v->playingPreset = -1;
}

static void tsf_voice_end(tsf* f, struct tsf_voice* v)
{
	// if maxVoiceNum is set, assume that voice rendering and note queuing are on separate threads
	// so to minimize the chance that voice rendering would advance the segment at the same time
	// we just do it twice here and hope that it sticks
	int repeats = (f->maxVoiceNum ? 2 : 1);
	while (repeats--)
	{
		tsf_voice_envelope_nextsegment(&v->ampenv, TSF_SEGMENT_SUSTAIN, f->outSampleRate);
		tsf_voice_envelope_nextsegment(&v->modenv, TSF_SEGMENT_SUSTAIN, f->outSampleRate);
		if (v->region->loop_mode == TSF_LOOPMODE_SUSTAIN)
		{
			// Continue playing, but stop looping.
			v->loopEnd = v->loopStart;
		}
	}
}

static void tsf_voice_endquick(tsf* f, struct tsf_voice* v)
{
	// if maxVoiceNum is set, assume that voice rendering and note queuing are on separate threads
	// so to minimize the chance that voice rendering would advance the segment at the same time
	// we just do it twice here and hope that it sticks
	int repeats = (f->maxVoiceNum ? 2 : 1);
	while (repeats--)
	{
		v->ampenv.parameters.release = 0.0f; tsf_voice_envelope_nextsegment(&v->ampenv, TSF_SEGMENT_SUSTAIN, f->outSampleRate);
		v->modenv.parameters.release = 0.0f; tsf_voice_envelope_nextsegment(&v->modenv, TSF_SEGMENT_SUSTAIN, f->outSampleRate);
	}
}

static void tsf_voice_calcpitchratio(struct tsf_voice* v, float pitchShift, float outSampleRate)
{
	double note = v->playingKey + v->region->transpose + v->region->tune / 100.0;
	double adjustedPitch = v->region->pitch_keycenter + (note - v->region->pitch_keycenter) * (v->region->pitch_keytrack / 100.0);
	if (pitchShift) adjustedPitch += pitchShift;
	v->pitchInputTimecents = adjustedPitch * 100.0;
	v->pitchOutputFactor = v->region->sample_rate / (tsf_timecents2Secsd(v->region->pitch_keycenter * 100.0) * outSampleRate);
}

static void tsf_voice_render(tsf* f, struct tsf_voice* v, float* outputBuffer, int numSamples)
{
	struct tsf_region* region = v->region;
	float* input = f->fontSamples;
	float* outL = outputBuffer;
	float* outR = (f->outputmode == TSF_STEREO_UNWEAVED ? outL + numSamples : TSF_NULL);

	// Cache some values, to give them at least some chance of ending up in registers.
	TSF_BOOL updateModEnv = (region->modEnvToPitch || region->modEnvToFilterFc);
	TSF_BOOL updateModLFO = (v->modlfo.delta && (region->modLfoToPitch || region->modLfoToFilterFc || region->modLfoToVolume));
	TSF_BOOL updateVibLFO = (v->viblfo.delta && (region->vibLfoToPitch));
	TSF_BOOL isLooping    = (v->loopStart < v->loopEnd);
	unsigned int tmpLoopStart = v->loopStart, tmpLoopEnd = v->loopEnd;
	double tmpSampleEndDbl = (double)region->end, tmpLoopEndDbl = (double)tmpLoopEnd + 1.0;
	double tmpSourceSamplePosition = v->sourceSamplePosition;
	struct tsf_voice_lowpass tmpLowpass = v->lowpass;

	TSF_BOOL dynamicLowpass = (region->modLfoToFilterFc || region->modEnvToFilterFc);
	float tmpSampleRate = f->outSampleRate, tmpInitialFilterFc, tmpModLfoToFilterFc, tmpModEnvToFilterFc;

	TSF_BOOL dynamicPitchRatio = (region->modLfoToPitch || region->modEnvToPitch || region->vibLfoToPitch);
	double pitchRatio;
	float tmpModLfoToPitch, tmpVibLfoToPitch, tmpModEnvToPitch;

	TSF_BOOL dynamicGain = (region->modLfoToVolume != 0);
	float noteGain = 0, tmpModLfoToVolume;

	if (dynamicLowpass) tmpInitialFilterFc = (float)region->initialFilterFc, tmpModLfoToFilterFc = (float)region->modLfoToFilterFc, tmpModEnvToFilterFc = (float)region->modEnvToFilterFc;
	else tmpInitialFilterFc = 0, tmpModLfoToFilterFc = 0, tmpModEnvToFilterFc = 0;

	if (dynamicPitchRatio) pitchRatio = 0, tmpModLfoToPitch = (float)region->modLfoToPitch, tmpVibLfoToPitch = (float)region->vibLfoToPitch, tmpModEnvToPitch = (float)region->modEnvToPitch;
	else pitchRatio = tsf_timecents2Secsd(v->pitchInputTimecents) * v->pitchOutputFactor, tmpModLfoToPitch = 0, tmpVibLfoToPitch = 0, tmpModEnvToPitch = 0;

	if (dynamicGain) tmpModLfoToVolume = (float)region->modLfoToVolume * 0.1f;
	else noteGain = tsf_decibelsToGain(v->noteGainDB), tmpModLfoToVolume = 0;

	while (numSamples)
	{
		float gainMono, gainLeft, gainRight;
		int blockSamples = (numSamples > TSF_RENDER_EFFECTSAMPLEBLOCK ? TSF_RENDER_EFFECTSAMPLEBLOCK : numSamples);
		numSamples -= blockSamples;