gsm_1215.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"/>
    <code><![CDATA[
      #include <stdlib.h>
      #include "ladspa-util.h"
      #include "gsm/gsm.h"
      #include "util/biquad.h"

      #define SCALE 32768.0f
      #define SCALE_R 0.0000305175f

      int bits[] = {0x01, 0x02, 0x04, 0x08, 0x10, 0x20, 0x40, 0x80};
    ]]></code>
  </global>

  <plugin label="gsm" id="1215" class="DistortionPlugin">
    <name>GSM simulator</name>
    <p>Encodes and decodes a signal using the GSM voice compression system. Has the effect of making the signal sound like it is being sent over a European mobile phone network.</p>

    <callback event="instantiate"><![CDATA[
      count = 0;
      resamp = s_rate / 8000;
      fs = s_rate;
      rsf = SCALE / (float)resamp;
      src = malloc(sizeof(gsm_signal) * 160);
      dst = malloc(sizeof(gsm_signal) * 163);
      dry = malloc(sizeof(LADSPA_Data) * 160 * resamp);
      handle = NULL;

      blf = malloc(sizeof(biquad));
      biquad_init(blf);
    ]]></callback>

    <callback event="activate"><![CDATA[
      count = 0;
      memset(src, 0, sizeof(gsm_signal) * 160);
      memset(dst, 0, sizeof(gsm_signal) * 163);
      memset(dry, 0, sizeof(LADSPA_Data) * 160 * resamp);
      handle = gsm_create();
      biquad_init(blf);
      hs_set_params(blf, 3500.0f, -50.0f, 0.7f, fs);
    ]]></callback>

    <callback event="cleanup"><![CDATA[
      free(plugin_data->src);
      free(plugin_data->dst);
      free(plugin_data->dry);
      free(plugin_data->blf);
      if (plugin_data->handle) {
        gsm_destroy(plugin_data->handle);
      }
    ]]></callback>

    <callback event="run" unused-vars="fs"><![CDATA[
      unsigned long pos;
      gsm_frame frame;
      int samp;
      float part;
      int error_rate = f_round(error);
      int num_passes = f_round(passes);

      for (pos = 0; pos < sample_count; pos++) {

	// oversample into buffer down to aprox 8kHz, 13bit
	src[count / resamp] += f_round(biquad_run(blf, input[pos]) * rsf);

	// interpolate output, so it doesn't sound totaly awful
	samp = count / resamp;
	part = (float)count / (float)resamp - (float)samp;
        buffer_write(output[pos], cube_interp(part, dst[samp], dst[samp+1], dst[samp+2], dst[samp+3]) * SCALE_R * drywet + dry[count] * (1.0f - drywet));

	// Maintain delayed, dry buffer.
	dry[count] = input[pos];

	count++;

	// If we have a full, downsampled buffer then run the encode +
	// decode process.
	if (count >= 160 * resamp) {
		int i, j;
		gsm_signal *in;

		count = 0;
		dst[0] = dst[160];
		dst[1] = dst[161];
		dst[2] = dst[162];

		in = src;
		for (j=0; j<num_passes; j++) {
			gsm_encode(handle, in, frame);
			for (i=0; i < error_rate; i++) {
				frame[1 + (rand() % 32)] ^= bits[rand() % 8];
			}
			gsm_decode(handle, frame, dst+3);
			in = dst+3;
		}
		if (num_passes == 0) {
			for (j=0; j < 160; j++) {
				dst[j + 3] = src[j];
			}
		}
		memset(src, 0, sizeof(gsm_signal) * 160);
	}
      }

      plugin_data->count = count;

      *(plugin_data->latency) = 160 * resamp;
    ]]></callback>

    <port label="drywet" dir="input" type="control" hint="default_1">
      <name>Dry/wet mix</name>
      <range min="0" max="1"/>
      <p>Controls the dry/wet mix, 0 will give you the dry signal (but with the appropriate amount of delay), 1 will give you a totally wet signal.</p>
    </port>

    <port label="passes" dir="input" type="control" hint="integer,default_1">
      <name>Number of passes</name>
      <range min="0" max="10"/>
      <p>The number of times the signal is sent through the encode/decode process. Increases the CPU consumption almost linearly, and it will become more peaky so less friendly to realtime processing.</p>
    </port>

    <port label="error" dir="input" type="control" hint="default_0">
      <name>Error rate (bits/block)</name>
      <range min="0" max="30"/>
      <p>The number of simulated bits that get changed during the transmission process.</p>
      <p>I really wanted to reduce the bandwidth to get that "shouting down a drainpipe" effect, but I'm not sure how the reduced bandwidth is dealt with by real phones. I suspect it's heavily patented technology.</p>
    </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="count" type="int" />
    <instance-data label="fs" type="float" />
    <instance-data label="rsf" type="float" />
    <instance-data label="resamp" type="int" />
    <instance-data label="handle" type="gsm" />
    <instance-data label="dry" type="LADSPA_Data *" />
    <instance-data label="src" type="gsm_signal *" />
    <instance-data label="dst" type="gsm_signal *" />
    <instance-data label="blf" type="biquad *" />
  </plugin>
</ladspa>