gnunet_gst.h File Reference

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Macros
#define	gst_element_factory_make(element, name)

Functions
void	pl_graph ()
GstElement *	gst_element_factory_make_debug (gchar *, gchar *)
	debug making elements More...
GstBin *	get_audiobin (GNUNET_gstData *, int)
GstBin *	get_coder (GNUNET_gstData *, int)
gboolean	gnunet_gst_bus_call (GstBus *bus, GstMessage *msg, gpointer data)
void	gg_setup_gst_bus (GNUNET_gstData *d)
void	gg_load_configuration (GNUNET_gstData *d)
GstFlowReturn	on_appsink_new_sample (GstElement *, GNUNET_gstData *)

Macro Definition Documentation

gst_element_factory_make

#define gst_element_factory_make	(		element,
			name
	)

Value:

    gst_element_factory_make_debug ( \
    element, name);

Definition at line 36 of file gnunet_gst.h.

Function Documentation

pl_graph()

void pl_graph

(

)

gst_element_factory_make_debug()

GstElement* gst_element_factory_make_debug	(	gchar *	factoryname,
		gchar *	name
	)

debug making elements

Definition at line 555 of file gnunet_gst.c.

{
  GstElement *element;
 
  element = gst_element_factory_make (factoryname, name);
 
  if (element == NULL)
  {
    printf ("\n Failed to create element - type: %s name: %s \n", factoryname,
            name);
    exit (10);
    return element;
  }
  else
  {
    return element;
  }
}

References gst_element_factory_make, and name.

get_audiobin()

GstBin* get_audiobin	(	GNUNET_gstData *	d,
		int	type
	)

Definition at line 950 of file gnunet_gst.c.

{
  GstBin *bin;
  GstElement *sink, *source, *queue, *conv, *resampler, *removesilence, *filter;
  GstPad *pad, *ghostpad;
  GstCaps *caps;
 
  if (type == SINK)
  {
    bin = GST_BIN (gst_bin_new ("Gnunet audiosink"));
 
    /* Create all the elements */
    if (d->dropsilence == TRUE)
    {
      queue = gst_element_factory_make ("queue", "queue");
      removesilence = gst_element_factory_make ("removesilence",
                                                "removesilence");
    }
 
    conv = gst_element_factory_make ("audioconvert", "converter");
    resampler = gst_element_factory_make ("audioresample", "resampler");
 
    if (d->audiobackend == AUTO)
    {
      sink = gst_element_factory_make ("autoaudiosink", "audiosink");
      g_signal_connect (sink, "child-added", G_CALLBACK (
                          autoaudiosink_child_added), NULL);
    }
 
    if (d->audiobackend == ALSA)
    {
      sink = gst_element_factory_make ("alsaaudiosink", "audiosink");
    }
 
    if (d->audiobackend == JACK)
    {
      sink = gst_element_factory_make ("jackaudiosink", "audiosink");
 
      g_object_set (G_OBJECT (sink), "client-name", "gnunet", NULL);
 
      if (g_object_class_find_property
            (G_OBJECT_GET_CLASS (sink), "port-pattern"))
      {
//        char *portpattern = "system";
 
        g_object_set (G_OBJECT (sink), "port-pattern", d->jack_pp_out,
                      NULL);
      }
    }
 
    if (d->audiobackend == FAKE)
    {
      sink = gst_element_factory_make ("fakesink", "audiosink");
    }
 
    g_object_set (sink,
                  "buffer-time", (gint64) BUFFER_TIME,
                  "latency-time", (gint64) LATENCY_TIME,
                  NULL);
 
    if (d->dropsilence == TRUE)
    {
      // Do not remove silence by default
      g_object_set (removesilence, "remove", FALSE, NULL);
      g_object_set (queue, "max-size-buffers", 12, NULL);
      /*
         g_signal_connect (source,
         "need-data",
         G_CALLBACK(appsrc_need_data),
         NULL);
 
         g_signal_connect (source,
         "enough-data",
         G_CALLBACK(appsrc_enough_data),
         NULL);
       *//*
      g_signal_connect (queue,
          "notify::current-level-bytes",
          G_CALLBACK(queue_current_level),
          NULL);
 
      g_signal_connect (queue,
          "underrun",
          G_CALLBACK(queue_underrun),
          NULL);
 
      g_signal_connect (queue,
          "running",
          G_CALLBACK(queue_running),
          NULL);
 
      g_signal_connect (queue,
          "overrun",
          G_CALLBACK(queue_overrun),
          NULL);
 
      g_signal_connect (queue,
          "pushing",
          G_CALLBACK(queue_pushing),
          NULL);
 */ }
 
 
    gst_bin_add_many (bin, conv, resampler, sink, NULL);
    gst_element_link_many (conv, resampler, sink, NULL);
 
    if (d->dropsilence == TRUE)
    {
      gst_bin_add_many (bin, queue, removesilence, NULL);
 
      if (! gst_element_link_many (queue, removesilence, conv, NULL))
        lf ("queue, removesilence, conv ");
 
      pad = gst_element_get_static_pad (queue, "sink");
    }
    else
    {
      pad = gst_element_get_static_pad (conv, "sink");
    }
 
    ghostpad = gst_ghost_pad_new ("sink", pad);
  }
  else
  {
    // SOURCE
 
    bin = GST_BIN (gst_bin_new ("Gnunet audiosource"));
 
    //    source = gst_element_factory_make("audiotestsrc", "audiotestsrcbla");
 
    if (d->audiobackend == AUTO)
    {
      source = gst_element_factory_make ("autoaudiosrc", "audiosource");
    }
    if (d->audiobackend == ALSA)
    {
      source = gst_element_factory_make ("alsasrc", "audiosource");
    }
    if (d->audiobackend == JACK)
    {
      source = gst_element_factory_make ("jackaudiosrc", "audiosource");
    }
    if (d->audiobackend == TEST)
    {
      source = gst_element_factory_make ("audiotestsrc", "audiosource");
    }
 
    filter = gst_element_factory_make ("capsfilter", "filter");
    conv = gst_element_factory_make ("audioconvert", "converter");
    resampler = gst_element_factory_make ("audioresample", "resampler");
 
    if (d->audiobackend == AUTO)
    {
      g_signal_connect (source, "child-added", G_CALLBACK (
                          autoaudiosource_child_added), NULL);
    }
    else
    {
      if (GST_IS_AUDIO_BASE_SRC (source))
        g_object_set (source, "buffer-time", (gint64) BUFFER_TIME,
                      "latency-time", (gint64) LATENCY_TIME, NULL);
      if (d->audiobackend == JACK)
      {
        g_object_set (G_OBJECT (source), "client-name", "gnunet", NULL);
        if (g_object_class_find_property
              (G_OBJECT_GET_CLASS (source), "port-pattern"))
        {
          char *portpattern = "moc";
 
          g_object_set (G_OBJECT (source), "port-pattern", portpattern,
                        NULL);
        }
      }
    }
 
    caps = gst_caps_new_simple ("audio/x-raw",
                                /*  "format", G_TYPE_STRING, "S16LE", */
                                /*    "rate", G_TYPE_INT, SAMPLING_RATE,*/
                                "channels", G_TYPE_INT, OPUS_CHANNELS,
                                /*    "layout", G_TYPE_STRING, "interleaved",*/
                                NULL);
 
    g_object_set (G_OBJECT (filter),
                  "caps", caps,
                  NULL);
    gst_caps_unref (caps);
 
    gst_bin_add_many (bin, source, filter, conv, resampler, NULL);
    gst_element_link_many (source, filter, conv, resampler, NULL);
 
    pad = gst_element_get_static_pad (resampler, "src");
 
 
    /* pads */
    ghostpad = gst_ghost_pad_new ("src", pad);
  }
 
  /* set the bin pads */
  gst_pad_set_active (ghostpad, TRUE);
  gst_element_add_pad (GST_ELEMENT (bin), ghostpad);
 
  gst_object_unref (pad);
 
  return bin;
}

References ALSA, GNUNET_gstData::audiobackend, AUTO, autoaudiosink_child_added(), autoaudiosource_child_added(), BUFFER_TIME, conv, GNUNET_gstData::dropsilence, FAKE, filter, gst_element_factory_make, JACK, GNUNET_gstData::jack_pp_out, LATENCY_TIME, lf(), OPUS_CHANNELS, queue(), resampler, sink, SINK, source, TEST, and type.

Referenced by main().

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get_coder()

GstBin* get_coder	(	GNUNET_gstData *	d,
		int	type
	)

Definition at line 816 of file gnunet_gst.c.

{
  GstBin *bin;
  GstPad *srcpad, *sinkpad, *srcghostpad, *sinkghostpad;
  GstCaps *rtpcaps;
  GstElement *encoder, *muxer, *decoder, *demuxer, *jitterbuffer,
             *rtpcapsfilter;
 
  if (d->usertp == TRUE)
  {
    /*
     * application/x-rtp, media=(string)audio, clock-rate=(int)48000, encoding-name=(string)OPUS, sprop-maxcapturerate=(string)48000, sprop-stereo=(string)0, payload=(int)96, encoding-params=(string)2, ssrc=(uint)630297634, timestamp-offset=(uint)678334141, seqnum-offset=(uint)16938 */
/*
    rtpcaps = gst_caps_new_simple ("application/x-rtp",
          "media", G_TYPE_STRING, "audio",
          "clock-rate", G_TYPE_INT, SAMPLING_RATE,
          "encoding-name", G_TYPE_STRING, "OPUS",
          "payload", G_TYPE_INT, 96,
          "sprop-stereo", G_TYPE_STRING, "0",
          "encoding-params", G_TYPE_STRING, "2",
          NULL);
 */ rtpcaps = gst_caps_new_simple ("application/x-rtp",
                                   "media", G_TYPE_STRING, "audio",
                                   "clock-rate", G_TYPE_INT, SAMPLING_RATE,
                                   "encoding-name", G_TYPE_STRING, "OPUS",
                                   "payload", G_TYPE_INT, 96,
                                   "sprop-stereo", G_TYPE_STRING, "0",
                                   "encoding-params", G_TYPE_STRING, "2",
                                   NULL);
 
 
    rtpcapsfilter = gst_element_factory_make ("capsfilter", "rtpcapsfilter");
 
    g_object_set (G_OBJECT (rtpcapsfilter),
                  "caps", rtpcaps,
                  NULL);
    gst_caps_unref (rtpcaps);
  }
 
 
  if (type == ENCODER)
  {
    bin = GST_BIN (gst_bin_new ("Gnunet audioencoder"));
 
    encoder = gst_element_factory_make ("opusenc", "opus-encoder");
    if (d->usertp == TRUE)
    {
      muxer = gst_element_factory_make ("rtpopuspay", "rtp-payloader");
    }
    else
    {
      muxer = gst_element_factory_make ("oggmux", "ogg-muxer");
    }
    g_object_set (G_OBJECT (encoder),
                  /*      "bitrate", 64000, */
                  /*      "bandwidth", OPUS_BANDWIDTH_FULLBAND, */
                  "inband-fec", INBAND_FEC_MODE,
                  "packet-loss-percentage", PACKET_LOSS_PERCENTAGE,
                  "max-payload-size", MAX_PAYLOAD_SIZE,
                  "audio", TRUE,  /* VoIP, not audio */
                  "frame-size", OPUS_FRAME_SIZE,
                  NULL);
 
    if (d->usertp != TRUE)
    {
      g_object_set (G_OBJECT (muxer),
                    "max-delay", OGG_MAX_DELAY,
                    "max-page-delay", OGG_MAX_PAGE_DELAY,
                    NULL);
    }
 
    gst_bin_add_many (bin, encoder, muxer, NULL);
    gst_element_link_many (encoder, muxer, NULL);
    sinkpad = gst_element_get_static_pad (encoder, "sink");
    sinkghostpad = gst_ghost_pad_new ("sink", sinkpad);
 
    srcpad = gst_element_get_static_pad (muxer, "src");
    srcghostpad = gst_ghost_pad_new ("src", srcpad);
  }
  if (type == DECODER)
  {
    bin = GST_BIN (gst_bin_new ("Gnunet audiodecoder"));
 
    // decoder
    if (d->usertp == TRUE)
    {
      demuxer = gst_element_factory_make ("rtpopusdepay", "ogg-demuxer");
      jitterbuffer = gst_element_factory_make ("rtpjitterbuffer",
                                               "rtpjitterbuffer");
    }
    else
    {
      demuxer = gst_element_factory_make ("oggdemux", "ogg-demuxer");
    }
    decoder = gst_element_factory_make ("opusdec", "opus-decoder");
 
    if (d->usertp == TRUE)
    {
      gst_bin_add_many (bin, rtpcapsfilter, jitterbuffer, demuxer, decoder,
                        NULL);
      gst_element_link_many (rtpcapsfilter, jitterbuffer, demuxer, decoder,
                             NULL);
      sinkpad = gst_element_get_static_pad (rtpcapsfilter, "sink");
    }
    else
    {
      gst_bin_add_many (bin, demuxer, decoder, NULL);
 
      g_signal_connect (demuxer,
                        "pad-added",
                        G_CALLBACK (decoder_ogg_pad_added),
                        decoder);
 
      sinkpad = gst_element_get_static_pad (demuxer, "sink");
    }
    sinkghostpad = gst_ghost_pad_new ("sink", sinkpad);
 
    srcpad = gst_element_get_static_pad (decoder, "src");
    srcghostpad = gst_ghost_pad_new ("src", srcpad);
  }
 
  // add pads to the bin
  gst_pad_set_active (sinkghostpad, TRUE);
  gst_element_add_pad (GST_ELEMENT (bin), sinkghostpad);
 
  gst_pad_set_active (srcghostpad, TRUE);
  gst_element_add_pad (GST_ELEMENT (bin), srcghostpad);
 
 
  return bin;
}

References decoder, DECODER, decoder_ogg_pad_added(), demuxer, ENCODER, gst_element_factory_make, INBAND_FEC_MODE, MAX_PAYLOAD_SIZE, OGG_MAX_DELAY, OGG_MAX_PAGE_DELAY, OPUS_FRAME_SIZE, PACKET_LOSS_PERCENTAGE, SAMPLING_RATE, type, and GNUNET_gstData::usertp.

Referenced by main().

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gnunet_gst_bus_call()

gboolean gnunet_gst_bus_call	(	GstBus *	bus,
		GstMessage *	msg,
		gpointer	data
	)

Definition at line 252 of file gnunet_gst.c.

{
  GNUNET_log (GNUNET_ERROR_TYPE_DEBUG,
              "Bus message\n");
  switch (GST_MESSAGE_TYPE (msg))
  {
  case GST_MESSAGE_EOS:
    GNUNET_log (GNUNET_ERROR_TYPE_INFO,
                "End of stream\n");
    exit (10);
    break;
 
  case GST_MESSAGE_ERROR:
    {
      gchar  *debug;
      GError *error;
 
      gst_message_parse_error (msg, &error, &debug);
      g_free (debug);
 
      GNUNET_log (GNUNET_ERROR_TYPE_ERROR,
                  "Error: %s\n",
                  error->message);
      g_error_free (error);
 
      exit (10);
      break;
    }
 
  default:
    break;
  }
 
  return TRUE;
}

References find_typedefs::debug, GNUNET_ERROR_TYPE_DEBUG, GNUNET_ERROR_TYPE_ERROR, GNUNET_ERROR_TYPE_INFO, GNUNET_log, and msg.

gg_setup_gst_bus()

void gg_setup_gst_bus

(

GNUNET_gstData *

d

)

Definition at line 355 of file gnunet_gst.c.

{
  GstBus *bus;
 
  bus = gst_element_get_bus (GST_ELEMENT (d->pipeline));
  gst_bus_add_signal_watch (bus);
  g_signal_connect (G_OBJECT (bus), "message::error", (GCallback) error_cb,
                    d);
  g_signal_connect (G_OBJECT (bus), "message::eos", (GCallback) eos_cb,
                    d);
  g_signal_connect (G_OBJECT (bus), "message::state-changed",
                    (GCallback) state_changed_cb, d);
  g_signal_connect (G_OBJECT (bus), "message::application",
                    (GCallback) application_cb, d);
  g_signal_connect (G_OBJECT (bus), "message::about-to-finish",
                    (GCallback) application_cb, d);
  gst_object_unref (bus);
}

References application_cb(), eos_cb(), error_cb(), GNUNET_gstData::pipeline, and state_changed_cb().

Referenced by main().

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gg_load_configuration()

void gg_load_configuration

(

GNUNET_gstData *

d

)

Definition at line 71 of file gnunet_gst.c.

{
  char *audiobackend_string;
 
  cfg = GNUNET_CONFIGURATION_create ();
  GNUNET_CONFIGURATION_load (cfg, "mediahelper.conf");
 
  GNUNET_CONFIGURATION_get_value_string (cfg, "MEDIAHELPER", "JACK_PP_IN",
                                         &d->jack_pp_in);
  GNUNET_CONFIGURATION_get_value_string (cfg, "MEDIAHELPER", "JACK_PP_OUT",
                                         &d->jack_pp_out);
 
  GNUNET_CONFIGURATION_get_value_string (cfg, "MEDIAHELPER", "AUDIOBACKEND",
                                         &audiobackend_string);
 
  // printf("abstring: %s \n", audiobackend_string);
 
  if (0 == strcasecmp (audiobackend_string, "AUTO"))
  {
    d->audiobackend = AUTO;
  }
  else if (0 == strcasecmp (audiobackend_string, "JACK"))
  {
    d->audiobackend = JACK;
  }
  else if (0 == strcasecmp (audiobackend_string, "ALSA"))
  {
    d->audiobackend = ALSA;
  }
  else if (0 == strcasecmp (audiobackend_string, "FAKE"))
  {
    d->audiobackend = FAKE;
  }
  else if (0 == strcasecmp (audiobackend_string, "TEST"))
  {
    d->audiobackend = TEST;
  }
  else
  {
    d->audiobackend = AUTO;
  }
 
  if (GNUNET_CONFIGURATION_get_value_yesno (cfg, "MEDIAHELPER",
                                            "REMOVESILENCE") == GNUNET_YES)
  {
    d->dropsilence = TRUE;
  }
  else
  {
    d->dropsilence = FALSE;
  }
 
  if (GNUNET_CONFIGURATION_get_value_yesno (cfg, "MEDIAHELPER",
                                            "NO_GN_HEADERS") == GNUNET_YES)
  {
    d->pure_ogg = TRUE;
  }
  else
  {
    d->pure_ogg = FALSE;
  }
 
 
  if (GNUNET_CONFIGURATION_get_value_yesno (cfg, "MEDIAHELPER", "USERTP") ==
      GNUNET_YES)
  {
    d->usertp = TRUE;
  }
  else
  {
    d->usertp = FALSE;
  }
 
//  GNUNET_CONFIGURATION_write(cfg, "mediahelper.conf");
}

References ALSA, GNUNET_gstData::audiobackend, AUTO, cfg, GNUNET_gstData::dropsilence, FAKE, GNUNET_CONFIGURATION_create(), GNUNET_CONFIGURATION_get_value_string(), GNUNET_CONFIGURATION_get_value_yesno(), GNUNET_CONFIGURATION_load(), GNUNET_YES, JACK, GNUNET_gstData::jack_pp_in, GNUNET_gstData::jack_pp_out, GNUNET_gstData::pure_ogg, TEST, and GNUNET_gstData::usertp.

Referenced by main().

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on_appsink_new_sample()

GstFlowReturn on_appsink_new_sample	(	GstElement *	element,
		GNUNET_gstData *	d
	)

Definition at line 170 of file gnunet_gst.c.

{
  // size of message including gnunet header
  size_t msg_size;
 
  GstSample *s;
  GstBuffer *b;
  GstMapInfo map;
 
/*
   const GstStructure *si;
   char *si_str;
   GstCaps *s_caps;
   char *caps_str;
 */if (gst_app_sink_is_eos (GST_APP_SINK (element)))
    return GST_FLOW_OK;
 
  // pull sample from appsink
  s = gst_app_sink_pull_sample (GST_APP_SINK (element));
 
  if (s == NULL)
    return GST_FLOW_OK;
 
  if (! GST_IS_SAMPLE (s))
    return GST_FLOW_OK;
 
  b = gst_sample_get_buffer (s);
 
  GST_WARNING ("caps are %" GST_PTR_FORMAT, gst_sample_get_caps (s));
 
 
  gst_buffer_map (b, &map, GST_MAP_READ);
 
  size_t len;
  len = map.size;
  if (len > UINT16_MAX - sizeof(struct AudioMessage))
  {
    // this should never happen?
    printf ("GSTREAMER sample too big! \n");
    exit (20);
    len = UINT16_MAX - sizeof(struct AudioMessage);
  }
 
  msg_size = sizeof(struct AudioMessage) + len;
 
  // copy the data into audio_message
  GNUNET_memcpy (((char *) &(d->audio_message)[1]), map.data, len);
  (d->audio_message)->header.size = htons ((uint16_t) msg_size);
  if (d->pure_ogg)
    // write the audio_message without the gnunet headers
    write_data ((const char *) &(d->audio_message)[1], len);
  else
    write_data ((const char *) d->audio_message, msg_size);
 
  gst_sample_unref (s);
  return GST_FLOW_OK;
}

References GNUNET_gstData::audio_message, GNUNET_memcpy, AudioMessage::header, len, map, GNUNET_gstData::pure_ogg, GNUNET_MessageHeader::size, GNUNET_CONTAINER_MultiPeerMap::size, and write_data().

Referenced by get_app().

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