audiofirfilter

## audiofirfilter audiofirfilter — Generic audio FIR filter with custom filter kernel |

## Synopsis

    struct              GstAudioFIRFilter;

## Object Hierarchy

      GObject
       +----GInitiallyUnowned
             +----GstObject
                   +----GstElement
                         +----GstBaseTransform
                               +----GstAudioFilter
                                     +----GstAudioFXBaseFIRFilter
                                           +----GstAudioFIRFilter

## Properties

      "kernel"                   GValueArray*          : Read / Write
      "latency"                  guint64               : Read / Write

## Signals

      "rate-changed"                                   : Run Last

## Description

audiofirfilter implements a generic audio FIR filter. Before usage the "kernel" property has to be set to the filter kernel that should be used and the "latency" property has to be set to the latency (in samples) that is introduced by the filter kernel. Setting a latency of n samples will lead to the first n samples being dropped from the output and n samples added to the end.

The filter kernel describes the impulse response of the filter. To calculate the frequency response of the filter you have to calculate the Fourier Transform of the impulse response.

To change the filter kernel whenever the sampling rate changes the "rate-changed" signal can be used. This should be done for most FIR filters as they're depending on the sampling rate.

### Example application

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    /* GStreamer
     * Copyright (C) 2009 Sebastian Droege <sebastian.droege@collabora.co.uk>
     *
     * This library is free software; you can redistribute it and/or
     * modify it under the terms of the GNU Library General Public
     * License as published by the Free Software Foundation; either
     * version 2 of the License, or (at your option) any later version.
     *
     * This library is distributed in the hope that it will be useful,
     * but WITHOUT ANY WARRANTY; without even the implied warranty of
     * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
     * Library General Public License for more details.
     *
     * You should have received a copy of the GNU Library General Public
     * License along with this library; if not, write to the
     * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
     * Boston, MA 02111-1307, USA.
     */

    /* This small sample application creates a bandpass FIR filter
     * by transforming the frequency response to the filter kernel.
     */

    /* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
     * with newer GLib versions (>= 2.31.0) */
    #define GLIB_DISABLE_DEPRECATION_WARNINGS

    #include <string.h>
    #include <math.h>

    #include <gst/gst.h>
    #include <gst/fft/gstfftf64.h>

    static gboolean
    on_message (GstBus * bus, GstMessage * message, gpointer user_data)
    {
      GMainLoop *loop = (GMainLoop *) user_data;

      switch (GST_MESSAGE_TYPE (message)) {
        case GST_MESSAGE_ERROR:
          g_error ("Got ERROR");
          g_main_loop_quit (loop);
          break;
        case GST_MESSAGE_WARNING:
          g_warning ("Got WARNING");
          g_main_loop_quit (loop);
          break;
        case GST_MESSAGE_EOS:
          g_main_loop_quit (loop);
          break;
        default:
          break;
      }

      return TRUE;
    }

    static void
    on_rate_changed (GstElement * element, gint rate, gpointer user_data)
    {
      GValueArray *va;
      GValue v = { 0, };
      GstFFTF64 *fft;
      GstFFTF64Complex frequency_response[17];
      gdouble tmp[32];
      gdouble filter_kernel[32];
      guint i;

      /* Create the frequency response: zero outside
       * a small frequency band */
      for (i = 0; i < 17; i++) {
        if (i < 5 || i > 11)
          frequency_response[i].r = 0.0;
        else
          frequency_response[i].r = 1.0;

        frequency_response[i].i = 0.0;
      }

      /* Calculate the inverse FT of the frequency response */
      fft = gst_fft_f64_new (32, TRUE);
      gst_fft_f64_inverse_fft (fft, frequency_response, tmp);
      gst_fft_f64_free (fft);

      /* Shift the inverse FT of the frequency response by 16,
       * i.e. the half of the kernel length to get the
       * impulse response. See http://www.dspguide.com/ch17/1.htm
       * for more information.
       */
      for (i = 0; i < 32; i++)
        filter_kernel[i] = tmp[(i + 16) % 32];

      /* Apply the hamming window to the impulse response to get
       * a better result than given from the rectangular window
       */
      for (i = 0; i < 32; i++)
        filter_kernel[i] *= (0.54 - 0.46 * cos (2 * G_PI * i / 32));

      va = g_value_array_new (1);

      g_value_init (&v, G_TYPE_DOUBLE);
      for (i = 0; i < 32; i++) {
        g_value_set_double (&v, filter_kernel[i]);
        g_value_array_append (va, &v);
        g_value_reset (&v);
      }
      g_object_set (G_OBJECT (element), "kernel", va, NULL);
      /* Latency is 1/2 of the kernel length for this method of
       * calculating a filter kernel from the frequency response
       */
      g_object_set (G_OBJECT (element), "latency", (gint64) (32 / 2), NULL);
      g_value_array_free (va);
    }

    gint
    main (gint argc, gchar * argv[])
    {
      GstElement *pipeline, *src, *filter, *conv, *sink;
      GstBus *bus;
      GMainLoop *loop;

      gst_init (NULL, NULL);

      pipeline = gst_element_factory_make ("pipeline", NULL);

      src = gst_element_factory_make ("audiotestsrc", NULL);
      g_object_set (G_OBJECT (src), "wave", 5, NULL);

      filter = gst_element_factory_make ("audiofirfilter", NULL);
      g_signal_connect (G_OBJECT (filter), "rate-changed",
          G_CALLBACK (on_rate_changed), NULL);

      conv = gst_element_factory_make ("audioconvert", NULL);

      sink = gst_element_factory_make ("autoaudiosink", NULL);
      g_return_val_if_fail (sink != NULL, -1);

      gst_bin_add_many (GST_BIN (pipeline), src, filter, conv, sink, NULL);
      if (!gst_element_link_many (src, filter, conv, sink, NULL)) {
        g_error ("Failed to link elements");
        return -2;
      }

      loop = g_main_loop_new (NULL, FALSE);

      bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
      gst_bus_add_signal_watch (bus);
      g_signal_connect (G_OBJECT (bus), "message", G_CALLBACK (on_message), loop);
      gst_object_unref (GST_OBJECT (bus));

      if (gst_element_set_state (pipeline,
              GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE) {
        g_error ("Failed to go into PLAYING state");
        return -3;
      }

      g_main_loop_run (loop);

      gst_element_set_state (pipeline, GST_STATE_NULL);

      g_main_loop_unref (loop);
      gst_object_unref (pipeline);

      return 0;
    }

## Synopsis

### Element Information

plugin | audiofx
author | Sebastian Dröge <sebastian.droege@collabora.co.uk>
class | Filter/Effect/Audio

### Element Pads

name | sink
direction | sink
presence | always
details | audio/x-raw, format=(string){ F32LE, F64LE }, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], layout=(string)interleaved

name | src
direction | source
presence | always
details | audio/x-raw, format=(string){ F32LE, F64LE }, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], layout=(string)interleaved

## Details

### struct GstAudioFIRFilter

    struct GstAudioFIRFilter;

Opaque data structure.

## Property Details

### The `"kernel"` property

      "kernel"                   GValueArray*          : Read / Write

Filter kernel for the FIR filter.

### The `"latency"` property

      "latency"                  guint64               : Read / Write

Filter latency in samples.

Default value: 0

## Signal Details

### The `"rate-changed"` signal

    void                user_function                      (GstAudioFIRFilter *filter,
                                                            gint               rate,
                                                            gpointer           user_data)      : Run Last

Will be emitted when the sampling rate changes. The callbacks will be called from the streaming thread and processing will stop until the event is handled.

`filter` : | the filter on which the signal is emitted
`rate` : | the new sampling rate
`user_data` : | user data set when the signal handler was connected.