pitch_scale_1193.xml

<?xml version="1.0" ?>
<!DOCTYPE ladspa SYSTEM "ladspa-swh.dtd">
<?xml-stylesheet href="ladspa.css" type="text/css" ?>
<ladspa>
	<global>
		<meta name="maker" value="Steve Harris &lt;steve@plugin.org.uk&gt;"/>
		<meta name="copyright" value="GPL"/>
		<meta name="properties" value="HARD_RT_CAPABLE"/>
		<code><![CDATA[
#include "util/pitchscale.h"

#define FRAME_LENGTH 2048
#define OVER_SAMP 4
		]]></code>
	</global>

	<plugin label="pitchScale" id="1193" class="PitchPlugin">
		<name>Pitch Scaler</name>
		<p>A pitch shifter implementation that scales the harmonics appropriately with the base frequencies. It is an implementation of Stephen M. Sprengler's pitch scaler design. It gives reasonable, general purpose results for small changes, but won't give Antares or Eventide anything to worry about.</p>
		<p>The FFT block size and oversampling has been kept at a minimum to keep the CPU usage low.</p>

		<callback event="run">
			pitch_scale(buffers, mult, FRAME_LENGTH, OVER_SAMP, sample_count, sample_rate, input, output, RUN_ADDING, run_adding_gain);

			*(plugin_data->latency) = FRAME_LENGTH - (FRAME_LENGTH /
							OVER_SAMP);
		</callback>

		<callback event="instantiate">
			int i;
			float arg;

			buffers = malloc(sizeof(sbuffers));
			sample_rate = s_rate;
			buffers->gInFIFO = malloc(FRAME_LENGTH * sizeof(float));
			buffers->gOutFIFO = malloc(FRAME_LENGTH * sizeof(float));
			buffers->gLastPhase = malloc(FRAME_LENGTH * sizeof(float));
			buffers->gSumPhase = malloc(FRAME_LENGTH * sizeof(float));
			buffers->gOutputAccum = malloc(2*FRAME_LENGTH * sizeof(float));
			buffers->gAnaFreq = malloc(FRAME_LENGTH * sizeof(float));
			buffers->gAnaMagn = malloc(FRAME_LENGTH * sizeof(float));
			buffers->gSynFreq = malloc(FRAME_LENGTH * sizeof(float));
			buffers->gSynMagn = malloc(FRAME_LENGTH * sizeof(float));
			buffers->gWindow = malloc(FRAME_LENGTH * sizeof(float));
/*
			if (aplan == NULL) {
#ifdef FFTW3
				aplan = fftwf_plan_r2r_1d(FRAME_LENGTH, ps_in, ps_out, FFTW_R2HC, FFTW_MEASURE);
				splan = fftwf_plan_r2r_1d(FRAME_LENGTH, ps_in, ps_out, FFTW_HC2R, FFTW_MEASURE);
#else
				aplan = rfftw_create_plan(FRAME_LENGTH, FFTW_REAL_TO_COMPLEX, FFTW_ESTIMATE);
				splan = rfftw_create_plan(FRAME_LENGTH, FFTW_COMPLEX_TO_REAL, FFTW_ESTIMATE);
#endif
			}
*/

			arg = 2.0f * M_PI / (float)(FRAME_LENGTH-1);
			for (i=0; i &lt; FRAME_LENGTH; i++) {
				// Blackman-Harris
				buffers->gWindow[i] =  0.35875f - 0.48829f * cos(arg * (float)i) + 0.14128f * cos(2.0f * arg * (float)i) - 0.01168f * cos(3.0f * arg * (float)i);
				// Gain correction
				buffers->gWindow[i] *= 0.761f;

			}
		</callback>

		<callback event="activate">
			memset(buffers->gInFIFO, 0, FRAME_LENGTH*sizeof(float));
			memset(buffers->gOutFIFO, 0, FRAME_LENGTH*sizeof(float));
			memset(buffers->gLastPhase, 0, FRAME_LENGTH*sizeof(float)/2);
			memset(buffers->gSumPhase, 0, FRAME_LENGTH*sizeof(float)/2);
			memset(buffers->gOutputAccum, 0, 2*FRAME_LENGTH*sizeof(float));
			memset(buffers->gAnaFreq, 0, FRAME_LENGTH*sizeof(float));
			memset(buffers->gAnaMagn, 0, FRAME_LENGTH*sizeof(float));
			buffers->gRover = 0;

			/* do one run to make sure the plans are set up */
			pitch_scale(buffers, 1.0, FRAME_LENGTH, 4, FRAME_LENGTH, sample_rate, buffers->gInFIFO, buffers->gOutFIFO, 0, 0.0f);
		</callback>

		<callback event="cleanup"><![CDATA[
			free (plugin_data->buffers->gInFIFO);
			free (plugin_data->buffers->gOutFIFO);
			free (plugin_data->buffers->gLastPhase);
			free (plugin_data->buffers->gSumPhase);
			free (plugin_data->buffers->gOutputAccum);
			free (plugin_data->buffers->gAnaFreq);
			free (plugin_data->buffers->gAnaMagn);
			free (plugin_data->buffers->gSynFreq);
			free (plugin_data->buffers->gSynMagn);
			free (plugin_data->buffers->gWindow);
			free (plugin_data->buffers);
		]]></callback>

		<port label="mult" dir="input" type="control" hint="default_1">
			<name>Pitch co-efficient</name>
			<range min="0.5" max="2"/>
		</port>

		<port label="input" dir="input" type="audio">
			<name>Input</name>
		</port>

		<port label="output" dir="output" type="audio">
			<name>Output</name>
		</port>

		<port label="latency" dir="output" type="control">
			<name>latency</name>
		</port>

		<instance-data label="buffers" type="sbuffers *"/>

		<instance-data label="sample_rate" type="long"/>
	</plugin>
</ladspa>