/* 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., 51 Franklin St, Fifth Floor,
 * Boston, MA 02110-1301, 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;
}

/* 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., 51 Franklin St, Fifth Floor,
 * Boston, MA 02110-1301, USA.
 */

/* This small sample application creates a lowpass IIR filter
 * and applies it to white noise.
 * See http://www.dspguide.com/ch19/2.htm for a description
 * of the IIR filter that is used.
 */

/* 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>

/* Cutoff of 4000 Hz */
#define CUTOFF (4000.0)

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, };
  gdouble x;

  if (rate / 2.0 > CUTOFF)
    x = exp (-2.0 * G_PI * (CUTOFF / rate));
  else
    x = 0.0;

  va = g_value_array_new (1);

  g_value_init (&v, G_TYPE_DOUBLE);
  g_value_set_double (&v, 1.0 - x);
  g_value_array_append (va, &v);
  g_value_reset (&v);
  g_object_set (G_OBJECT (element), "a", va, NULL);
  g_value_array_free (va);

  va = g_value_array_new (1);
  g_value_set_double (&v, x);
  g_value_array_append (va, &v);
  g_value_reset (&v);
  g_object_set (G_OBJECT (element), "b", va, 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 ("audioiirfilter", 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;
}

/* GStreamer
 * Copyright (C) 2000,2001,2002,2003,2005
 *           Thomas Vander Stichele <thomas at apestaart dot org>
 *
 * 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., 51 Franklin St, Fifth Floor,
 * Boston, MA 02110-1301, USA.
 */

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

#define GLIB_DISABLE_DEPRECATION_WARNINGS

#include <gst/gst.h>

static gboolean
message_handler (GstBus * bus, GstMessage * message, gpointer data)
{

  if (message->type == GST_MESSAGE_ELEMENT) {
    const GstStructure *s = gst_message_get_structure (message);
    const gchar *name = gst_structure_get_name (s);

    if (strcmp (name, "level") == 0) {
      gint channels;
      GstClockTime endtime;
      gdouble rms_dB, peak_dB, decay_dB;
      gdouble rms;
      const GValue *array_val;
      const GValue *value;
      GValueArray *rms_arr, *peak_arr, *decay_arr;
      gint i;

      if (!gst_structure_get_clock_time (s, "endtime", &endtime))
        g_warning ("Could not parse endtime");

      /* the values are packed into GValueArrays with the value per channel */
      array_val = gst_structure_get_value (s, "rms");
      rms_arr = (GValueArray *) g_value_get_boxed (array_val);

      array_val = gst_structure_get_value (s, "peak");
      peak_arr = (GValueArray *) g_value_get_boxed (array_val);

      array_val = gst_structure_get_value (s, "decay");
      decay_arr = (GValueArray *) g_value_get_boxed (array_val);

      /* we can get the number of channels as the length of any of the value
       * arrays */
      channels = rms_arr->n_values;
      g_print ("endtime: %" GST_TIME_FORMAT ", channels: %d\n",
          GST_TIME_ARGS (endtime), channels);
      for (i = 0; i < channels; ++i) {

        g_print ("channel %d\n", i);
        value = g_value_array_get_nth (rms_arr, i);
        rms_dB = g_value_get_double (value);

        value = g_value_array_get_nth (peak_arr, i);
        peak_dB = g_value_get_double (value);

        value = g_value_array_get_nth (decay_arr, i);
        decay_dB = g_value_get_double (value);
        g_print ("    RMS: %f dB, peak: %f dB, decay: %f dB\n",
            rms_dB, peak_dB, decay_dB);

        /* converting from dB to normal gives us a value between 0.0 and 1.0 */
        rms = pow (10, rms_dB / 20);
        g_print ("    normalized rms value: %f\n", rms);
      }
    }
  }
  /* we handled the message we want, and ignored the ones we didn't want.
   * so the core can unref the message for us */
  return TRUE;
}

int
main (int argc, char *argv[])
{
  GstElement *audiotestsrc, *audioconvert, *level, *fakesink;
  GstElement *pipeline;
  GstCaps *caps;
  GstBus *bus;
  guint watch_id;
  GMainLoop *loop;

  gst_init (&argc, &argv);

  caps = gst_caps_from_string ("audio/x-raw,channels=2");

  pipeline = gst_pipeline_new (NULL);
  g_assert (pipeline);
  audiotestsrc = gst_element_factory_make ("audiotestsrc", NULL);
  g_assert (audiotestsrc);
  audioconvert = gst_element_factory_make ("audioconvert", NULL);
  g_assert (audioconvert);
  level = gst_element_factory_make ("level", NULL);
  g_assert (level);
  fakesink = gst_element_factory_make ("fakesink", NULL);
  g_assert (fakesink);

  gst_bin_add_many (GST_BIN (pipeline), audiotestsrc, audioconvert, level,
      fakesink, NULL);
  if (!gst_element_link (audiotestsrc, audioconvert))
    g_error ("Failed to link audiotestsrc and audioconvert");
  if (!gst_element_link_filtered (audioconvert, level, caps))
    g_error ("Failed to link audioconvert and level");
  if (!gst_element_link (level, fakesink))
    g_error ("Failed to link level and fakesink");

  /* make sure we'll get messages */
  g_object_set (G_OBJECT (level), "post-messages", TRUE, NULL);
  /* run synced and not as fast as we can */
  g_object_set (G_OBJECT (fakesink), "sync", TRUE, NULL);

  bus = gst_element_get_bus (pipeline);
  watch_id = gst_bus_add_watch (bus, message_handler, NULL);

  gst_element_set_state (pipeline, GST_STATE_PLAYING);

  /* we need to run a GLib main loop to get the messages */
  loop = g_main_loop_new (NULL, FALSE);
  g_main_loop_run (loop);

  g_source_remove (watch_id);
  g_main_loop_unref (loop);
  return 0;
}

/* GStreamer
 * Copyright (C) 2006 Stefan Kost <ensonic@users.sf.net>
 * Copyright (C) 2008 Jan Schmidt <jan.schmidt@sun.com>
 *
 * 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., 51 Franklin St, Fifth Floor,
 * Boston, MA 02110-1301, USA.
 */

#include <stdio.h>
#include <string.h>
#include <stdlib.h>
#include <gst/gst.h>

static guint spect_bands = 20;

#define AUDIOFREQ 32000

/* receive spectral data from element message */
static gboolean
message_handler (GstBus * bus, GstMessage * message, gpointer data)
{
  if (message->type == GST_MESSAGE_ELEMENT) {
    const GstStructure *s = gst_message_get_structure (message);
    const gchar *name = gst_structure_get_name (s);
    GstClockTime endtime;

    if (strcmp (name, "spectrum") == 0) {
      const GValue *magnitudes;
      const GValue *phases;
      const GValue *mag, *phase;
      gdouble freq;
      guint i;

      if (!gst_structure_get_clock_time (s, "endtime", &endtime))
        endtime = GST_CLOCK_TIME_NONE;

      g_print ("New spectrum message, endtime %" GST_TIME_FORMAT "\n",
          GST_TIME_ARGS (endtime));

      magnitudes = gst_structure_get_value (s, "magnitude");
      phases = gst_structure_get_value (s, "phase");

      for (i = 0; i < spect_bands; ++i) {
        freq = (gdouble) ((AUDIOFREQ / 2) * i + AUDIOFREQ / 4) / spect_bands;
        mag = gst_value_list_get_value (magnitudes, i);
        phase = gst_value_list_get_value (phases, i);

        if (mag != NULL && phase != NULL) {
          g_print ("band %d (freq %g): magnitude %f dB phase %f\n", i, freq,
              g_value_get_float (mag), g_value_get_float (phase));
        }
      }
      g_print ("\n");
    }
  }
  return TRUE;
}

int
main (int argc, char *argv[])
{
  GstElement *bin;
  GstElement *src, *audioconvert, *spectrum, *sink;
  GstBus *bus;
  GstCaps *caps;
  GMainLoop *loop;

  gst_init (&argc, &argv);

  bin = gst_pipeline_new ("bin");

  src = gst_element_factory_make ("audiotestsrc", "src");
  g_object_set (G_OBJECT (src), "wave", 0, "freq", 6000.0, NULL);
  audioconvert = gst_element_factory_make ("audioconvert", NULL);
  g_assert (audioconvert);

  spectrum = gst_element_factory_make ("spectrum", "spectrum");
  g_object_set (G_OBJECT (spectrum), "bands", spect_bands, "threshold", -80,
      "post-messages", TRUE, "message-phase", TRUE, NULL);

  sink = gst_element_factory_make ("fakesink", "sink");
  g_object_set (G_OBJECT (sink), "sync", TRUE, NULL);

  gst_bin_add_many (GST_BIN (bin), src, audioconvert, spectrum, sink, NULL);

  caps = gst_caps_new_simple ("audio/x-raw",
      "rate", G_TYPE_INT, AUDIOFREQ, NULL);

  if (!gst_element_link (src, audioconvert) ||
      !gst_element_link_filtered (audioconvert, spectrum, caps) ||
      !gst_element_link (spectrum, sink)) {
    fprintf (stderr, "can't link elements\n");
    exit (1);
  }
  gst_caps_unref (caps);

  bus = gst_element_get_bus (bin);
  gst_bus_add_watch (bus, message_handler, NULL);
  gst_object_unref (bus);

  gst_element_set_state (bin, GST_STATE_PLAYING);

  /* we need to run a GLib main loop to get the messages */
  loop = g_main_loop_new (NULL, FALSE);
  g_main_loop_run (loop);

  gst_element_set_state (bin, GST_STATE_NULL);

  gst_object_unref (bin);

  return 0;
}