Problem using appsink on simple example

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Problem using appsink on simple example

Tiago Katcipis
Im going to use appsink on a project using gstreamer and started to build a basic example os]f how to use appsink using signals. My problem is that when i simply add the appsink to the pipeline, the pipe simply stops to work.

On the following example im just trying to use the appsink instead the filesink to write the entire stream to a file. If i use the filesink and DONT add the appsink to the pipe, it works fine, if i just add the appsink to the pipe ....it stops working, im not even using the appsink yet, the filesink stops to write data, the result of the test will be an empty file, and no error msg is sent. I dont know what detail im missing on how to use appsink, hope someone can help me. By the way, gst_bin_add returns TRUE when i add the appsink.

the source code of the example:

#include <gst/gst.h>
#include <glib.h>
#include <gst/app/gstappsink.h>
#include <stdio.h>

static gboolean
bus_call (GstBus     *bus,
          GstMessage *msg,
          gpointer    data)
{
  GMainLoop *loop = (GMainLoop *) data;

  switch (GST_MESSAGE_TYPE (msg)) {

    case GST_MESSAGE_EOS:
      g_print ("End of stream\n");
      g_main_loop_quit (loop);
      break;

    case GST_MESSAGE_ERROR: {
      gchar  *debug;
      GError *error;

      gst_message_parse_error (msg, &error, &debug);
      g_free (debug);

      g_printerr ("Error: %s\n", error->message);
      g_error_free (error);

      g_main_loop_quit (loop);
      break;
    }
    default:
      g_print("Msg type[%d], Msg type name[%s]\n", GST_MESSAGE_TYPE(msg), GST_MESSAGE_TYPE_NAME(msg));
      break;
  }

  return TRUE;
}


static void link_two_elements(GstElement* src_element, GstElement* sink_element)
{
  if(!gst_element_link(src_element, sink_element))
      g_debug("Error linking %s to %s", gst_element_get_name(src_element), gst_element_get_name(sink_element));
 
}

static void link_two_pads(GstPad* src_pad, GstPad* sink_pad)
{
  if(!src_pad){
      g_warning("Error: src_pad is NULL on link_two_pads");
      return;
  }

  if(!sink_pad){
      g_warning("Error: sink_pad is NULL on link_two_pads");
      return;
  }

  if(gst_pad_link(src_pad, sink_pad) != GST_PAD_LINK_OK)
      g_debug("Error linking pads %s to %s", gst_pad_get_name(src_pad), gst_pad_get_name(sink_pad));

}

static void on_new_buffer (GstElement* object,
                           gpointer user_data)
{
  FILE* file = (FILE*) user_data;
  GstAppSink* app_sink = (GstAppSink*) object;
  GstBuffer * buffer = gst_app_sink_pull_buffer(app_sink);

  if(fwrite (GST_BUFFER_DATA(buffer), 1 , GST_BUFFER_SIZE(buffer) , file) != GST_BUFFER_SIZE(buffer)){
      g_debug("Error writing data from appsink to file!!!");
  }else{
      g_debug("Data pulled from appsink and writed to file with success!!");
  }
 
}

// Pipe to test this src: gst-launch audiotestsrc ! audioconvert ! alawenc ! rtppcmapay ! udpsink host=127.0.0.1 port=5000
// Equivalent working pipe: gst-launch udpsrc port=5000 caps=application/x-rtp ! gstrtpjitterbuffer ! rtppcmadepay ! alawdec ! audioconvert ! lame ! appsink
int
main (int   argc,
      char *argv[])
{
  GMainLoop *loop;

  GstElement *pipeline, *source, *rtp_jitter, *rtp_alaw_depay,
             *alaw_decoder, *audio_convert, *lame, *filesink, *appsink;
  GstBus* bus;
  GstCaps* udp_caps;
  FILE* appsink_file;
  int udp_port;

  if(argc < 2){
      g_warning("Usage: %s [port_to_be_listened]", argv[0]);
      return -1;
  }

  udp_port = atoi(argv[1]);

  /* Initialisation */
  gst_init (&argc, &argv);

  udp_caps = gst_caps_from_string("application/x-rtp");
  if(!udp_caps){
      g_warning("Error alocating the udp caps");
      return -1;
  }

  loop = g_main_loop_new (NULL, FALSE);

  /* Create gstreamer elements */
  pipeline      = gst_pipeline_new("rtp-mp3-stream-decoder");
  source        = gst_element_factory_make("udpsrc",  "udp-rtp-source");
  rtp_jitter     = gst_element_factory_make("gstrtpjitterbuffer",  "rtp-jitter-buffer");
  rtp_alaw_depay = gst_element_factory_make("rtppcmadepay",  "rtp_alaw_depay");
  alaw_decoder   = gst_element_factory_make("alawdec","alaw-decoder");
  audio_convert  = gst_element_factory_make("audioconvert","audio-convert");
  lame           = gst_element_factory_make("lame","mp3-encoder");
  filesink       = gst_element_factory_make("filesink", "file-mp3-output");
  appsink        = gst_element_factory_make("appsink", "sink-buffer");
 

  if (!pipeline || !source || !rtp_jitter || !appsink ||
      !rtp_alaw_depay || !alaw_decoder || !audio_convert || !lame || !filesink) {
      g_printerr ("Elements could not be created. Exiting.\n");
      return -1;
  }

  appsink_file = fopen("received_audio_appsink.mp3", "w");
  if(!appsink_file){
      g_printerr ("Appsink file could not be created. Exiting.\n");
      return -1;
  }

  /* Set up the pipeline */

  /* we set the properties to the source element to receive only rtp packets*/
  g_object_set(G_OBJECT (source), "port", udp_port, NULL);
  g_object_set(G_OBJECT (source), "caps", udp_caps, NULL);
  /* we set the location of the mp3 generated file */
  g_object_set(G_OBJECT (filesink), "location", "received_audio_filesink.mp3", NULL);

  /*
    Make appsink emit the "new-preroll" and "new-buffer" signals. This option is by default disabled because
    signal emission is expensive and unneeded when the application prefers to operate in pull mode.
  */
  gst_app_sink_set_emit_signals ((GstAppSink*) appsink, TRUE);

  /* we add a message handler */
  bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
  gst_bus_add_watch (bus, bus_call, loop);
  gst_object_unref (bus);

  /* we add all elements into the pipeline */
  gst_bin_add_many (GST_BIN (pipeline),
                    source, rtp_jitter, rtp_alaw_depay, alaw_decoder, audio_convert, lame, filesink, NULL);

  /* When i just addn the appsink this the example stops to work and the file will be empty (and im not using the appsink, just the filesink)
  if(gst_bin_add(GST_BIN (pipeline), appsink)){
      g_debug("Adcionou appsink com sucesso");
  }else{
      g_debug("Erro ao Adcionar appsink");
  }*/

  /* we link all the elements together */
  link_two_elements(source, rtp_jitter);
  link_two_elements(rtp_jitter, rtp_alaw_depay);
  link_two_elements(rtp_alaw_depay, alaw_decoder);
  link_two_elements(alaw_decoder, audio_convert);
  link_two_elements(audio_convert, lame);
  link_two_elements(lame, filesink);

  /* Conecting to the new-buffer signal emited by the appsink */
  g_signal_connect (appsink, "new-buffer",  G_CALLBACK (on_new_buffer), appsink_file);

  /* Set the pipeline to "playing" state*/
  g_print ("Now listening on port: %d\n", udp_port);
  gst_element_set_state (pipeline, GST_STATE_PLAYING);

  /* Iterate */
  g_print ("Running...\n");
  g_main_loop_run (loop);

  /* Out of the main loop, clean up nicely */
  g_print ("Returned, stopping listening\n");
  gst_element_set_state (pipeline, GST_STATE_NULL);

  g_print ("Deleting pipeline\n");
  gst_object_unref (GST_OBJECT (pipeline));
  fclose(appsink_file);

  return 0;
}



best regards,
Katcipis
--
"it might be a profitable thing to learn Java, but it has no intellectual value whatsoever" Alexander Stepanov

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Re: Problem using appsink on simple example

Tim-Philipp Müller-2
On Wed, 2009-07-08 at 11:25 -0300, Tiago Katcipis wrote:

> If i use the filesink and DONT add the appsink to the pipe, it works
> fine. If i just add the appsink to the pipe ....it stops working, im
> not even using the appsink yet, the filesink stops to write data, the
> result of the test will be an empty file, and no error msg is sent.

This is somewhat expected. The pipeline is unlikely to preroll (go to
>=PAUSED state) if there's an unconnected sink element in it, since it
will wait for that sink to receive data, but if it's not connected the
sink will never receive any data.

Cheers
 -Tim



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Re: Problem using appsink on simple example

Thiago Sousa Santos-2
In reply to this post by Tiago Katcipis


On Wed, Jul 8, 2009 at 11:25 AM, Tiago Katcipis <[hidden email]> wrote:
Im going to use appsink on a project using gstreamer and started to build a basic example os]f how to use appsink using signals. My problem is that when i simply add the appsink to the pipeline, the pipe simply stops to work.

On the following example im just trying to use the appsink instead the filesink to write the entire stream to a file. If i use the filesink and DONT add the appsink to the pipe, it works fine, if i just add the appsink to the pipe ....it stops working, im not even using the appsink yet, the filesink stops to write data, the result of the test will be an empty file, and no error msg is sent. I dont know what detail im missing on how to use appsink, hope someone can help me. By the way, gst_bin_add returns TRUE when i add the appsink.

the source code of the example:

#include <gst/gst.h>
#include <glib.h>
#include <gst/app/gstappsink.h>
#include <stdio.h>

static gboolean
bus_call (GstBus     *bus,
          GstMessage *msg,
          gpointer    data)
{
  GMainLoop *loop = (GMainLoop *) data;

  switch (GST_MESSAGE_TYPE (msg)) {

    case GST_MESSAGE_EOS:
      g_print ("End of stream\n");
      g_main_loop_quit (loop);
      break;

    case GST_MESSAGE_ERROR: {
      gchar  *debug;
      GError *error;

      gst_message_parse_error (msg, &error, &debug);
      g_free (debug);

      g_printerr ("Error: %s\n", error->message);
      g_error_free (error);

      g_main_loop_quit (loop);
      break;
    }
    default:
      g_print("Msg type[%d], Msg type name[%s]\n", GST_MESSAGE_TYPE(msg), GST_MESSAGE_TYPE_NAME(msg));
      break;
  }

  return TRUE;
}


static void link_two_elements(GstElement* src_element, GstElement* sink_element)
{
  if(!gst_element_link(src_element, sink_element))
      g_debug("Error linking %s to %s", gst_element_get_name(src_element), gst_element_get_name(sink_element));
 
}

static void link_two_pads(GstPad* src_pad, GstPad* sink_pad)
{
  if(!src_pad){
      g_warning("Error: src_pad is NULL on link_two_pads");
      return;
  }

  if(!sink_pad){
      g_warning("Error: sink_pad is NULL on link_two_pads");
      return;
  }

  if(gst_pad_link(src_pad, sink_pad) != GST_PAD_LINK_OK)
      g_debug("Error linking pads %s to %s", gst_pad_get_name(src_pad), gst_pad_get_name(sink_pad));

}

static void on_new_buffer (GstElement* object,
                           gpointer user_data)
{
  FILE* file = (FILE*) user_data;
  GstAppSink* app_sink = (GstAppSink*) object;
  GstBuffer * buffer = gst_app_sink_pull_buffer(app_sink);

  if(fwrite (GST_BUFFER_DATA(buffer), 1 , GST_BUFFER_SIZE(buffer) , file) != GST_BUFFER_SIZE(buffer)){
      g_debug("Error writing data from appsink to file!!!");
  }else{
      g_debug("Data pulled from appsink and writed to file with success!!");
  }
 
}

// Pipe to test this src: gst-launch audiotestsrc ! audioconvert ! alawenc ! rtppcmapay ! udpsink host=127.0.0.1 port=5000
// Equivalent working pipe: gst-launch udpsrc port=5000 caps=application/x-rtp ! gstrtpjitterbuffer ! rtppcmadepay ! alawdec ! audioconvert ! lame ! appsink
int
main (int   argc,
      char *argv[])
{
  GMainLoop *loop;

  GstElement *pipeline, *source, *rtp_jitter, *rtp_alaw_depay,
             *alaw_decoder, *audio_convert, *lame, *filesink, *appsink;
  GstBus* bus;
  GstCaps* udp_caps;
  FILE* appsink_file;
  int udp_port;

  if(argc < 2){
      g_warning("Usage: %s [port_to_be_listened]", argv[0]);
      return -1;
  }

  udp_port = atoi(argv[1]);

  /* Initialisation */
  gst_init (&argc, &argv);

  udp_caps = gst_caps_from_string("application/x-rtp");
  if(!udp_caps){
      g_warning("Error alocating the udp caps");
      return -1;
  }

  loop = g_main_loop_new (NULL, FALSE);

  /* Create gstreamer elements */
  pipeline      = gst_pipeline_new("rtp-mp3-stream-decoder");
  source        = gst_element_factory_make("udpsrc",  "udp-rtp-source");
  rtp_jitter     = gst_element_factory_make("gstrtpjitterbuffer",  "rtp-jitter-buffer");
  rtp_alaw_depay = gst_element_factory_make("rtppcmadepay",  "rtp_alaw_depay");
  alaw_decoder   = gst_element_factory_make("alawdec","alaw-decoder");
  audio_convert  = gst_element_factory_make("audioconvert","audio-convert");
  lame           = gst_element_factory_make("lame","mp3-encoder");
  filesink       = gst_element_factory_make("filesink", "file-mp3-output");
  appsink        = gst_element_factory_make("appsink", "sink-buffer");
 

  if (!pipeline || !source || !rtp_jitter || !appsink ||
      !rtp_alaw_depay || !alaw_decoder || !audio_convert || !lame || !filesink) {
      g_printerr ("Elements could not be created. Exiting.\n");
      return -1;
  }

  appsink_file = fopen("received_audio_appsink.mp3", "w");
  if(!appsink_file){
      g_printerr ("Appsink file could not be created. Exiting.\n");
      return -1;
  }

  /* Set up the pipeline */

  /* we set the properties to the source element to receive only rtp packets*/
  g_object_set(G_OBJECT (source), "port", udp_port, NULL);
  g_object_set(G_OBJECT (source), "caps", udp_caps, NULL);
  /* we set the location of the mp3 generated file */
  g_object_set(G_OBJECT (filesink), "location", "received_audio_filesink.mp3", NULL);

  /*
    Make appsink emit the "new-preroll" and "new-buffer" signals. This option is by default disabled because
    signal emission is expensive and unneeded when the application prefers to operate in pull mode.
  */
  gst_app_sink_set_emit_signals ((GstAppSink*) appsink, TRUE);

  /* we add a message handler */
  bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
  gst_bus_add_watch (bus, bus_call, loop);
  gst_object_unref (bus);

  /* we add all elements into the pipeline */
  gst_bin_add_many (GST_BIN (pipeline),
                    source, rtp_jitter, rtp_alaw_depay, alaw_decoder, audio_convert, lame, filesink, NULL);

  /* When i just addn the appsink this the example stops to work and the file will be empty (and im not using the appsink, just the filesink)
  if(gst_bin_add(GST_BIN (pipeline), appsink)){
      g_debug("Adcionou appsink com sucesso");
  }else{
      g_debug("Erro ao Adcionar appsink");
  }*/

  /* we link all the elements together */
  link_two_elements(source, rtp_jitter);
  link_two_elements(rtp_jitter, rtp_alaw_depay);
  link_two_elements(rtp_alaw_depay, alaw_decoder);
  link_two_elements(alaw_decoder, audio_convert);
  link_two_elements(audio_convert, lame);
  link_two_elements(lame, filesink);

Appsink (in this case of yours) is meant to *replace* filesink. You don't seem to link appsink to anything and also you link filesink to lame. So, remove filesink from your pipeline, and use appsink in its place.
 


  /* Conecting to the new-buffer signal emited by the appsink */
  g_signal_connect (appsink, "new-buffer",  G_CALLBACK (on_new_buffer), appsink_file);

  /* Set the pipeline to "playing" state*/
  g_print ("Now listening on port: %d\n", udp_port);
  gst_element_set_state (pipeline, GST_STATE_PLAYING);

  /* Iterate */
  g_print ("Running...\n");
  g_main_loop_run (loop);

  /* Out of the main loop, clean up nicely */
  g_print ("Returned, stopping listening\n");
  gst_element_set_state (pipeline, GST_STATE_NULL);

  g_print ("Deleting pipeline\n");
  gst_object_unref (GST_OBJECT (pipeline));
  fclose(appsink_file);

  return 0;
}



best regards,
Katcipis
--
"it might be a profitable thing to learn Java, but it has no intellectual value whatsoever" Alexander Stepanov

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--
Thiago Sousa Santos

Embedded Systems and Pervasive Computing Lab (Embedded)
Center of Electrical Engineering and Informatics (CEEI)
Federal University of Campina Grande (UFCG)

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Re: Problem using appsink on simple example

Tiago Katcipis
Puting appsink on the place of filesink dont works too. My problem is that even filesink stops to work when i just add the appsink to the bin. It seens that only adding the appsink to the bin makes the pipe stop working.

best regards,
Katcipis

On Wed, Jul 8, 2009 at 7:32 PM, thiagoss <[hidden email]> wrote:


On Wed, Jul 8, 2009 at 11:25 AM, Tiago Katcipis <[hidden email]> wrote:
Im going to use appsink on a project using gstreamer and started to build a basic example os]f how to use appsink using signals. My problem is that when i simply add the appsink to the pipeline, the pipe simply stops to work.

On the following example im just trying to use the appsink instead the filesink to write the entire stream to a file. If i use the filesink and DONT add the appsink to the pipe, it works fine, if i just add the appsink to the pipe ....it stops working, im not even using the appsink yet, the filesink stops to write data, the result of the test will be an empty file, and no error msg is sent. I dont know what detail im missing on how to use appsink, hope someone can help me. By the way, gst_bin_add returns TRUE when i add the appsink.

the source code of the example:

#include <gst/gst.h>
#include <glib.h>
#include <gst/app/gstappsink.h>
#include <stdio.h>

static gboolean
bus_call (GstBus     *bus,
          GstMessage *msg,
          gpointer    data)
{
  GMainLoop *loop = (GMainLoop *) data;

  switch (GST_MESSAGE_TYPE (msg)) {

    case GST_MESSAGE_EOS:
      g_print ("End of stream\n");
      g_main_loop_quit (loop);
      break;

    case GST_MESSAGE_ERROR: {
      gchar  *debug;
      GError *error;

      gst_message_parse_error (msg, &error, &debug);
      g_free (debug);

      g_printerr ("Error: %s\n", error->message);
      g_error_free (error);

      g_main_loop_quit (loop);
      break;
    }
    default:
      g_print("Msg type[%d], Msg type name[%s]\n", GST_MESSAGE_TYPE(msg), GST_MESSAGE_TYPE_NAME(msg));
      break;
  }

  return TRUE;
}


static void link_two_elements(GstElement* src_element, GstElement* sink_element)
{
  if(!gst_element_link(src_element, sink_element))
      g_debug("Error linking %s to %s", gst_element_get_name(src_element), gst_element_get_name(sink_element));
 
}

static void link_two_pads(GstPad* src_pad, GstPad* sink_pad)
{
  if(!src_pad){
      g_warning("Error: src_pad is NULL on link_two_pads");
      return;
  }

  if(!sink_pad){
      g_warning("Error: sink_pad is NULL on link_two_pads");
      return;
  }

  if(gst_pad_link(src_pad, sink_pad) != GST_PAD_LINK_OK)
      g_debug("Error linking pads %s to %s", gst_pad_get_name(src_pad), gst_pad_get_name(sink_pad));

}

static void on_new_buffer (GstElement* object,
                           gpointer user_data)
{
  FILE* file = (FILE*) user_data;
  GstAppSink* app_sink = (GstAppSink*) object;
  GstBuffer * buffer = gst_app_sink_pull_buffer(app_sink);

  if(fwrite (GST_BUFFER_DATA(buffer), 1 , GST_BUFFER_SIZE(buffer) , file) != GST_BUFFER_SIZE(buffer)){
      g_debug("Error writing data from appsink to file!!!");
  }else{
      g_debug("Data pulled from appsink and writed to file with success!!");
  }
 
}

// Pipe to test this src: gst-launch audiotestsrc ! audioconvert ! alawenc ! rtppcmapay ! udpsink host=127.0.0.1 port=5000
// Equivalent working pipe: gst-launch udpsrc port=5000 caps=application/x-rtp ! gstrtpjitterbuffer ! rtppcmadepay ! alawdec ! audioconvert ! lame ! appsink
int
main (int   argc,
      char *argv[])
{
  GMainLoop *loop;

  GstElement *pipeline, *source, *rtp_jitter, *rtp_alaw_depay,
             *alaw_decoder, *audio_convert, *lame, *filesink, *appsink;
  GstBus* bus;
  GstCaps* udp_caps;
  FILE* appsink_file;
  int udp_port;

  if(argc < 2){
      g_warning("Usage: %s [port_to_be_listened]", argv[0]);
      return -1;
  }

  udp_port = atoi(argv[1]);

  /* Initialisation */
  gst_init (&argc, &argv);

  udp_caps = gst_caps_from_string("application/x-rtp");
  if(!udp_caps){
      g_warning("Error alocating the udp caps");
      return -1;
  }

  loop = g_main_loop_new (NULL, FALSE);

  /* Create gstreamer elements */
  pipeline      = gst_pipeline_new("rtp-mp3-stream-decoder");
  source        = gst_element_factory_make("udpsrc",  "udp-rtp-source");
  rtp_jitter     = gst_element_factory_make("gstrtpjitterbuffer",  "rtp-jitter-buffer");
  rtp_alaw_depay = gst_element_factory_make("rtppcmadepay",  "rtp_alaw_depay");
  alaw_decoder   = gst_element_factory_make("alawdec","alaw-decoder");
  audio_convert  = gst_element_factory_make("audioconvert","audio-convert");
  lame           = gst_element_factory_make("lame","mp3-encoder");
  filesink       = gst_element_factory_make("filesink", "file-mp3-output");
  appsink        = gst_element_factory_make("appsink", "sink-buffer");
 

  if (!pipeline || !source || !rtp_jitter || !appsink ||
      !rtp_alaw_depay || !alaw_decoder || !audio_convert || !lame || !filesink) {
      g_printerr ("Elements could not be created. Exiting.\n");
      return -1;
  }

  appsink_file = fopen("received_audio_appsink.mp3", "w");
  if(!appsink_file){
      g_printerr ("Appsink file could not be created. Exiting.\n");
      return -1;
  }

  /* Set up the pipeline */

  /* we set the properties to the source element to receive only rtp packets*/
  g_object_set(G_OBJECT (source), "port", udp_port, NULL);
  g_object_set(G_OBJECT (source), "caps", udp_caps, NULL);
  /* we set the location of the mp3 generated file */
  g_object_set(G_OBJECT (filesink), "location", "received_audio_filesink.mp3", NULL);

  /*
    Make appsink emit the "new-preroll" and "new-buffer" signals. This option is by default disabled because
    signal emission is expensive and unneeded when the application prefers to operate in pull mode.
  */
  gst_app_sink_set_emit_signals ((GstAppSink*) appsink, TRUE);

  /* we add a message handler */
  bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
  gst_bus_add_watch (bus, bus_call, loop);
  gst_object_unref (bus);

  /* we add all elements into the pipeline */
  gst_bin_add_many (GST_BIN (pipeline),
                    source, rtp_jitter, rtp_alaw_depay, alaw_decoder, audio_convert, lame, filesink, NULL);

  /* When i just addn the appsink this the example stops to work and the file will be empty (and im not using the appsink, just the filesink)
  if(gst_bin_add(GST_BIN (pipeline), appsink)){
      g_debug("Adcionou appsink com sucesso");
  }else{
      g_debug("Erro ao Adcionar appsink");
  }*/

  /* we link all the elements together */
  link_two_elements(source, rtp_jitter);
  link_two_elements(rtp_jitter, rtp_alaw_depay);
  link_two_elements(rtp_alaw_depay, alaw_decoder);
  link_two_elements(alaw_decoder, audio_convert);
  link_two_elements(audio_convert, lame);
  link_two_elements(lame, filesink);

Appsink (in this case of yours) is meant to *replace* filesink. You don't seem to link appsink to anything and also you link filesink to lame. So, remove filesink from your pipeline, and use appsink in its place.
 


  /* Conecting to the new-buffer signal emited by the appsink */
  g_signal_connect (appsink, "new-buffer",  G_CALLBACK (on_new_buffer), appsink_file);

  /* Set the pipeline to "playing" state*/
  g_print ("Now listening on port: %d\n", udp_port);
  gst_element_set_state (pipeline, GST_STATE_PLAYING);

  /* Iterate */
  g_print ("Running...\n");
  g_main_loop_run (loop);

  /* Out of the main loop, clean up nicely */
  g_print ("Returned, stopping listening\n");
  gst_element_set_state (pipeline, GST_STATE_NULL);

  g_print ("Deleting pipeline\n");
  gst_object_unref (GST_OBJECT (pipeline));
  fclose(appsink_file);

  return 0;
}



best regards,
Katcipis
--
"it might be a profitable thing to learn Java, but it has no intellectual value whatsoever" Alexander Stepanov

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--
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Embedded Systems and Pervasive Computing Lab (Embedded)
Center of Electrical Engineering and Informatics (CEEI)
Federal University of Campina Grande (UFCG)

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Re: Problem using appsink on simple example

Andoni Morales
2009/7/9 Tiago Katcipis <[hidden email]>:
> Puting appsink on the place of filesink dont works too. My problem is that
> even filesink stops to work when i just add the appsink to the bin. It seens
If you need to have an element in the pipeline that it's not linked
b/c you may want to link it dinamically you need to lock it's state
usin GST_STATE_LOCK(appsink), this way state changes won't affect to
this element.
Andoni

> that only adding the appsink to the bin makes the pipe stop working.
>
> best regards,
> Katcipis
>
> On Wed, Jul 8, 2009 at 7:32 PM, thiagoss <[hidden email]> wrote:
>>
>>
>> On Wed, Jul 8, 2009 at 11:25 AM, Tiago Katcipis <[hidden email]>
>> wrote:
>>>
>>> Im going to use appsink on a project using gstreamer and started to build
>>> a basic example os]f how to use appsink using signals. My problem is that
>>> when i simply add the appsink to the pipeline, the pipe simply stops to
>>> work.
>>>
>>> On the following example im just trying to use the appsink instead the
>>> filesink to write the entire stream to a file. If i use the filesink and
>>> DONT add the appsink to the pipe, it works fine, if i just add the appsink
>>> to the pipe ....it stops working, im not even using the appsink yet, the
>>> filesink stops to write data, the result of the test will be an empty file,
>>> and no error msg is sent. I dont know what detail im missing on how to use
>>> appsink, hope someone can help me. By the way, gst_bin_add returns TRUE when
>>> i add the appsink.
>>>
>>> the source code of the example:
>>>
>>> #include <gst/gst.h>
>>> #include <glib.h>
>>> #include <gst/app/gstappsink.h>
>>> #include <stdio.h>
>>>
>>> static gboolean
>>> bus_call (GstBus     *bus,
>>>           GstMessage *msg,
>>>           gpointer    data)
>>> {
>>>   GMainLoop *loop = (GMainLoop *) data;
>>>
>>>   switch (GST_MESSAGE_TYPE (msg)) {
>>>
>>>     case GST_MESSAGE_EOS:
>>>       g_print ("End of stream\n");
>>>       g_main_loop_quit (loop);
>>>       break;
>>>
>>>     case GST_MESSAGE_ERROR: {
>>>       gchar  *debug;
>>>       GError *error;
>>>
>>>       gst_message_parse_error (msg, &error, &debug);
>>>       g_free (debug);
>>>
>>>       g_printerr ("Error: %s\n", error->message);
>>>       g_error_free (error);
>>>
>>>       g_main_loop_quit (loop);
>>>       break;
>>>     }
>>>     default:
>>>       g_print("Msg type[%d], Msg type name[%s]\n", GST_MESSAGE_TYPE(msg),
>>> GST_MESSAGE_TYPE_NAME(msg));
>>>       break;
>>>   }
>>>
>>>   return TRUE;
>>> }
>>>
>>>
>>> static void link_two_elements(GstElement* src_element, GstElement*
>>> sink_element)
>>> {
>>>   if(!gst_element_link(src_element, sink_element))
>>>       g_debug("Error linking %s to %s",
>>> gst_element_get_name(src_element), gst_element_get_name(sink_element));
>>>
>>> }
>>>
>>> static void link_two_pads(GstPad* src_pad, GstPad* sink_pad)
>>> {
>>>   if(!src_pad){
>>>       g_warning("Error: src_pad is NULL on link_two_pads");
>>>       return;
>>>   }
>>>
>>>   if(!sink_pad){
>>>       g_warning("Error: sink_pad is NULL on link_two_pads");
>>>       return;
>>>   }
>>>
>>>   if(gst_pad_link(src_pad, sink_pad) != GST_PAD_LINK_OK)
>>>       g_debug("Error linking pads %s to %s", gst_pad_get_name(src_pad),
>>> gst_pad_get_name(sink_pad));
>>>
>>> }
>>>
>>> static void on_new_buffer (GstElement* object,
>>>                            gpointer user_data)
>>> {
>>>   FILE* file = (FILE*) user_data;
>>>   GstAppSink* app_sink = (GstAppSink*) object;
>>>   GstBuffer * buffer = gst_app_sink_pull_buffer(app_sink);
>>>
>>>   if(fwrite (GST_BUFFER_DATA(buffer), 1 , GST_BUFFER_SIZE(buffer) , file)
>>> != GST_BUFFER_SIZE(buffer)){
>>>       g_debug("Error writing data from appsink to file!!!");
>>>   }else{
>>>       g_debug("Data pulled from appsink and writed to file with
>>> success!!");
>>>   }
>>>
>>> }
>>>
>>> // Pipe to test this src: gst-launch audiotestsrc ! audioconvert !
>>> alawenc ! rtppcmapay ! udpsink host=127.0.0.1 port=5000
>>> // Equivalent working pipe: gst-launch udpsrc port=5000
>>> caps=application/x-rtp ! gstrtpjitterbuffer ! rtppcmadepay ! alawdec !
>>> audioconvert ! lame ! appsink
>>> int
>>> main (int   argc,
>>>       char *argv[])
>>> {
>>>   GMainLoop *loop;
>>>
>>>   GstElement *pipeline, *source, *rtp_jitter, *rtp_alaw_depay,
>>>              *alaw_decoder, *audio_convert, *lame, *filesink, *appsink;
>>>   GstBus* bus;
>>>   GstCaps* udp_caps;
>>>   FILE* appsink_file;
>>>   int udp_port;
>>>
>>>   if(argc < 2){
>>>       g_warning("Usage: %s [port_to_be_listened]", argv[0]);
>>>       return -1;
>>>   }
>>>
>>>   udp_port = atoi(argv[1]);
>>>
>>>   /* Initialisation */
>>>   gst_init (&argc, &argv);
>>>
>>>   udp_caps = gst_caps_from_string("application/x-rtp");
>>>   if(!udp_caps){
>>>       g_warning("Error alocating the udp caps");
>>>       return -1;
>>>   }
>>>
>>>   loop = g_main_loop_new (NULL, FALSE);
>>>
>>>   /* Create gstreamer elements */
>>>   pipeline      = gst_pipeline_new("rtp-mp3-stream-decoder");
>>>   source        = gst_element_factory_make("udpsrc",  "udp-rtp-source");
>>>   rtp_jitter     = gst_element_factory_make("gstrtpjitterbuffer",
>>> "rtp-jitter-buffer");
>>>   rtp_alaw_depay = gst_element_factory_make("rtppcmadepay",
>>> "rtp_alaw_depay");
>>>   alaw_decoder   = gst_element_factory_make("alawdec","alaw-decoder");
>>>   audio_convert  =
>>> gst_element_factory_make("audioconvert","audio-convert");
>>>   lame           = gst_element_factory_make("lame","mp3-encoder");
>>>   filesink       = gst_element_factory_make("filesink",
>>> "file-mp3-output");
>>>   appsink        = gst_element_factory_make("appsink", "sink-buffer");
>>>
>>>
>>>   if (!pipeline || !source || !rtp_jitter || !appsink ||
>>>       !rtp_alaw_depay || !alaw_decoder || !audio_convert || !lame ||
>>> !filesink) {
>>>       g_printerr ("Elements could not be created. Exiting.\n");
>>>       return -1;
>>>   }
>>>
>>>   appsink_file = fopen("received_audio_appsink.mp3", "w");
>>>   if(!appsink_file){
>>>       g_printerr ("Appsink file could not be created. Exiting.\n");
>>>       return -1;
>>>   }
>>>
>>>   /* Set up the pipeline */
>>>
>>>   /* we set the properties to the source element to receive only rtp
>>> packets*/
>>>   g_object_set(G_OBJECT (source), "port", udp_port, NULL);
>>>   g_object_set(G_OBJECT (source), "caps", udp_caps, NULL);
>>>   /* we set the location of the mp3 generated file */
>>>   g_object_set(G_OBJECT (filesink), "location",
>>> "received_audio_filesink.mp3", NULL);
>>>
>>>   /*
>>>     Make appsink emit the "new-preroll" and "new-buffer" signals. This
>>> option is by default disabled because
>>>     signal emission is expensive and unneeded when the application
>>> prefers to operate in pull mode.
>>>   */
>>>   gst_app_sink_set_emit_signals ((GstAppSink*) appsink, TRUE);
>>>
>>>   /* we add a message handler */
>>>   bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
>>>   gst_bus_add_watch (bus, bus_call, loop);
>>>   gst_object_unref (bus);
>>>
>>>   /* we add all elements into the pipeline */
>>>   gst_bin_add_many (GST_BIN (pipeline),
>>>                     source, rtp_jitter, rtp_alaw_depay, alaw_decoder,
>>> audio_convert, lame, filesink, NULL);
>>>
>>>   /* When i just addn the appsink this the example stops to work and the
>>> file will be empty (and im not using the appsink, just the filesink)
>>>   if(gst_bin_add(GST_BIN (pipeline), appsink)){
>>>       g_debug("Adcionou appsink com sucesso");
>>>   }else{
>>>       g_debug("Erro ao Adcionar appsink");
>>>   }*/
>>>
>>>   /* we link all the elements together */
>>>   link_two_elements(source, rtp_jitter);
>>>   link_two_elements(rtp_jitter, rtp_alaw_depay);
>>>   link_two_elements(rtp_alaw_depay, alaw_decoder);
>>>   link_two_elements(alaw_decoder, audio_convert);
>>>   link_two_elements(audio_convert, lame);
>>>   link_two_elements(lame, filesink);
>>
>> Appsink (in this case of yours) is meant to *replace* filesink. You don't
>> seem to link appsink to anything and also you link filesink to lame. So,
>> remove filesink from your pipeline, and use appsink in its place.
>>
>>>
>>>   /* Conecting to the new-buffer signal emited by the appsink */
>>>   g_signal_connect (appsink, "new-buffer",  G_CALLBACK (on_new_buffer),
>>> appsink_file);
>>>
>>>   /* Set the pipeline to "playing" state*/
>>>   g_print ("Now listening on port: %d\n", udp_port);
>>>   gst_element_set_state (pipeline, GST_STATE_PLAYING);
>>>
>>>   /* Iterate */
>>>   g_print ("Running...\n");
>>>   g_main_loop_run (loop);
>>>
>>>   /* Out of the main loop, clean up nicely */
>>>   g_print ("Returned, stopping listening\n");
>>>   gst_element_set_state (pipeline, GST_STATE_NULL);
>>>
>>>   g_print ("Deleting pipeline\n");
>>>   gst_object_unref (GST_OBJECT (pipeline));
>>>   fclose(appsink_file);
>>>
>>>   return 0;
>>> }
>>>
>>>
>>>
>>> best regards,
>>> Katcipis
>>> --
>>> "it might be a profitable thing to learn Java, but it has no intellectual
>>> value whatsoever" Alexander Stepanov
>>>
>>>
>>> ------------------------------------------------------------------------------
>>> Enter the BlackBerry Developer Challenge
>>> This is your chance to win up to $100,000 in prizes! For a limited time,
>>> vendors submitting new applications to BlackBerry App World(TM) will have
>>> the opportunity to enter the BlackBerry Developer Challenge. See full
>>> prize
>>> details at: http://p.sf.net/sfu/Challenge
>>> _______________________________________________
>>> gstreamer-devel mailing list
>>> [hidden email]
>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>>>
>>
>>
>>
>> --
>> Thiago Sousa Santos
>>
>> Embedded Systems and Pervasive Computing Lab (Embedded)
>> Center of Electrical Engineering and Informatics (CEEI)
>> Federal University of Campina Grande (UFCG)
>>
>>
>> ------------------------------------------------------------------------------
>> Enter the BlackBerry Developer Challenge
>> This is your chance to win up to $100,000 in prizes! For a limited time,
>> vendors submitting new applications to BlackBerry App World(TM) will have
>> the opportunity to enter the BlackBerry Developer Challenge. See full
>> prize
>> details at: http://p.sf.net/sfu/Challenge
>> _______________________________________________
>> gstreamer-devel mailing list
>> [hidden email]
>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>>
>
>
>
> --
> "it might be a profitable thing to learn Java, but it has no intellectual
> value whatsoever" Alexander Stepanov
>
> ------------------------------------------------------------------------------
> Enter the BlackBerry Developer Challenge
> This is your chance to win up to $100,000 in prizes! For a limited time,
> vendors submitting new applications to BlackBerry App World(TM) will have
> the opportunity to enter the BlackBerry Developer Challenge. See full prize
> details at: http://p.sf.net/sfu/Challenge
> _______________________________________________
> gstreamer-devel mailing list
> [hidden email]
> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>
>



--
Andoni Morales Alastruey

LongoMatch:The Digital Coach
http://www.longomatch.ylatuya.es

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Re: Problem using appsink on simple example

Andoni Morales
2009/7/9 Andoni Morales <[hidden email]>:
> 2009/7/9 Tiago Katcipis <[hidden email]>:
>> Puting appsink on the place of filesink dont works too. My problem is that
>> even filesink stops to work when i just add the appsink to the bin. It seens
> If you need to have an element in the pipeline that it's not linked
> b/c you may want to link it dinamically you need to lock it's state
> usin GST_STATE_LOCK(appsink), this way state changes won't affect to
I meant gst_element_set_locked_state(appsink)

> this element.
> Andoni
>> that only adding the appsink to the bin makes the pipe stop working.
>>
>> best regards,
>> Katcipis
>>
>> On Wed, Jul 8, 2009 at 7:32 PM, thiagoss <[hidden email]> wrote:
>>>
>>>
>>> On Wed, Jul 8, 2009 at 11:25 AM, Tiago Katcipis <[hidden email]>
>>> wrote:
>>>>
>>>> Im going to use appsink on a project using gstreamer and started to build
>>>> a basic example os]f how to use appsink using signals. My problem is that
>>>> when i simply add the appsink to the pipeline, the pipe simply stops to
>>>> work.
>>>>
>>>> On the following example im just trying to use the appsink instead the
>>>> filesink to write the entire stream to a file. If i use the filesink and
>>>> DONT add the appsink to the pipe, it works fine, if i just add the appsink
>>>> to the pipe ....it stops working, im not even using the appsink yet, the
>>>> filesink stops to write data, the result of the test will be an empty file,
>>>> and no error msg is sent. I dont know what detail im missing on how to use
>>>> appsink, hope someone can help me. By the way, gst_bin_add returns TRUE when
>>>> i add the appsink.
>>>>
>>>> the source code of the example:
>>>>
>>>> #include <gst/gst.h>
>>>> #include <glib.h>
>>>> #include <gst/app/gstappsink.h>
>>>> #include <stdio.h>
>>>>
>>>> static gboolean
>>>> bus_call (GstBus     *bus,
>>>>           GstMessage *msg,
>>>>           gpointer    data)
>>>> {
>>>>   GMainLoop *loop = (GMainLoop *) data;
>>>>
>>>>   switch (GST_MESSAGE_TYPE (msg)) {
>>>>
>>>>     case GST_MESSAGE_EOS:
>>>>       g_print ("End of stream\n");
>>>>       g_main_loop_quit (loop);
>>>>       break;
>>>>
>>>>     case GST_MESSAGE_ERROR: {
>>>>       gchar  *debug;
>>>>       GError *error;
>>>>
>>>>       gst_message_parse_error (msg, &error, &debug);
>>>>       g_free (debug);
>>>>
>>>>       g_printerr ("Error: %s\n", error->message);
>>>>       g_error_free (error);
>>>>
>>>>       g_main_loop_quit (loop);
>>>>       break;
>>>>     }
>>>>     default:
>>>>       g_print("Msg type[%d], Msg type name[%s]\n", GST_MESSAGE_TYPE(msg),
>>>> GST_MESSAGE_TYPE_NAME(msg));
>>>>       break;
>>>>   }
>>>>
>>>>   return TRUE;
>>>> }
>>>>
>>>>
>>>> static void link_two_elements(GstElement* src_element, GstElement*
>>>> sink_element)
>>>> {
>>>>   if(!gst_element_link(src_element, sink_element))
>>>>       g_debug("Error linking %s to %s",
>>>> gst_element_get_name(src_element), gst_element_get_name(sink_element));
>>>>
>>>> }
>>>>
>>>> static void link_two_pads(GstPad* src_pad, GstPad* sink_pad)
>>>> {
>>>>   if(!src_pad){
>>>>       g_warning("Error: src_pad is NULL on link_two_pads");
>>>>       return;
>>>>   }
>>>>
>>>>   if(!sink_pad){
>>>>       g_warning("Error: sink_pad is NULL on link_two_pads");
>>>>       return;
>>>>   }
>>>>
>>>>   if(gst_pad_link(src_pad, sink_pad) != GST_PAD_LINK_OK)
>>>>       g_debug("Error linking pads %s to %s", gst_pad_get_name(src_pad),
>>>> gst_pad_get_name(sink_pad));
>>>>
>>>> }
>>>>
>>>> static void on_new_buffer (GstElement* object,
>>>>                            gpointer user_data)
>>>> {
>>>>   FILE* file = (FILE*) user_data;
>>>>   GstAppSink* app_sink = (GstAppSink*) object;
>>>>   GstBuffer * buffer = gst_app_sink_pull_buffer(app_sink);
>>>>
>>>>   if(fwrite (GST_BUFFER_DATA(buffer), 1 , GST_BUFFER_SIZE(buffer) , file)
>>>> != GST_BUFFER_SIZE(buffer)){
>>>>       g_debug("Error writing data from appsink to file!!!");
>>>>   }else{
>>>>       g_debug("Data pulled from appsink and writed to file with
>>>> success!!");
>>>>   }
>>>>
>>>> }
>>>>
>>>> // Pipe to test this src: gst-launch audiotestsrc ! audioconvert !
>>>> alawenc ! rtppcmapay ! udpsink host=127.0.0.1 port=5000
>>>> // Equivalent working pipe: gst-launch udpsrc port=5000
>>>> caps=application/x-rtp ! gstrtpjitterbuffer ! rtppcmadepay ! alawdec !
>>>> audioconvert ! lame ! appsink
>>>> int
>>>> main (int   argc,
>>>>       char *argv[])
>>>> {
>>>>   GMainLoop *loop;
>>>>
>>>>   GstElement *pipeline, *source, *rtp_jitter, *rtp_alaw_depay,
>>>>              *alaw_decoder, *audio_convert, *lame, *filesink, *appsink;
>>>>   GstBus* bus;
>>>>   GstCaps* udp_caps;
>>>>   FILE* appsink_file;
>>>>   int udp_port;
>>>>
>>>>   if(argc < 2){
>>>>       g_warning("Usage: %s [port_to_be_listened]", argv[0]);
>>>>       return -1;
>>>>   }
>>>>
>>>>   udp_port = atoi(argv[1]);
>>>>
>>>>   /* Initialisation */
>>>>   gst_init (&argc, &argv);
>>>>
>>>>   udp_caps = gst_caps_from_string("application/x-rtp");
>>>>   if(!udp_caps){
>>>>       g_warning("Error alocating the udp caps");
>>>>       return -1;
>>>>   }
>>>>
>>>>   loop = g_main_loop_new (NULL, FALSE);
>>>>
>>>>   /* Create gstreamer elements */
>>>>   pipeline      = gst_pipeline_new("rtp-mp3-stream-decoder");
>>>>   source        = gst_element_factory_make("udpsrc",  "udp-rtp-source");
>>>>   rtp_jitter     = gst_element_factory_make("gstrtpjitterbuffer",
>>>> "rtp-jitter-buffer");
>>>>   rtp_alaw_depay = gst_element_factory_make("rtppcmadepay",
>>>> "rtp_alaw_depay");
>>>>   alaw_decoder   = gst_element_factory_make("alawdec","alaw-decoder");
>>>>   audio_convert  =
>>>> gst_element_factory_make("audioconvert","audio-convert");
>>>>   lame           = gst_element_factory_make("lame","mp3-encoder");
>>>>   filesink       = gst_element_factory_make("filesink",
>>>> "file-mp3-output");
>>>>   appsink        = gst_element_factory_make("appsink", "sink-buffer");
>>>>
>>>>
>>>>   if (!pipeline || !source || !rtp_jitter || !appsink ||
>>>>       !rtp_alaw_depay || !alaw_decoder || !audio_convert || !lame ||
>>>> !filesink) {
>>>>       g_printerr ("Elements could not be created. Exiting.\n");
>>>>       return -1;
>>>>   }
>>>>
>>>>   appsink_file = fopen("received_audio_appsink.mp3", "w");
>>>>   if(!appsink_file){
>>>>       g_printerr ("Appsink file could not be created. Exiting.\n");
>>>>       return -1;
>>>>   }
>>>>
>>>>   /* Set up the pipeline */
>>>>
>>>>   /* we set the properties to the source element to receive only rtp
>>>> packets*/
>>>>   g_object_set(G_OBJECT (source), "port", udp_port, NULL);
>>>>   g_object_set(G_OBJECT (source), "caps", udp_caps, NULL);
>>>>   /* we set the location of the mp3 generated file */
>>>>   g_object_set(G_OBJECT (filesink), "location",
>>>> "received_audio_filesink.mp3", NULL);
>>>>
>>>>   /*
>>>>     Make appsink emit the "new-preroll" and "new-buffer" signals. This
>>>> option is by default disabled because
>>>>     signal emission is expensive and unneeded when the application
>>>> prefers to operate in pull mode.
>>>>   */
>>>>   gst_app_sink_set_emit_signals ((GstAppSink*) appsink, TRUE);
>>>>
>>>>   /* we add a message handler */
>>>>   bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
>>>>   gst_bus_add_watch (bus, bus_call, loop);
>>>>   gst_object_unref (bus);
>>>>
>>>>   /* we add all elements into the pipeline */
>>>>   gst_bin_add_many (GST_BIN (pipeline),
>>>>                     source, rtp_jitter, rtp_alaw_depay, alaw_decoder,
>>>> audio_convert, lame, filesink, NULL);
>>>>
>>>>   /* When i just addn the appsink this the example stops to work and the
>>>> file will be empty (and im not using the appsink, just the filesink)
>>>>   if(gst_bin_add(GST_BIN (pipeline), appsink)){
>>>>       g_debug("Adcionou appsink com sucesso");
>>>>   }else{
>>>>       g_debug("Erro ao Adcionar appsink");
>>>>   }*/
>>>>
>>>>   /* we link all the elements together */
>>>>   link_two_elements(source, rtp_jitter);
>>>>   link_two_elements(rtp_jitter, rtp_alaw_depay);
>>>>   link_two_elements(rtp_alaw_depay, alaw_decoder);
>>>>   link_two_elements(alaw_decoder, audio_convert);
>>>>   link_two_elements(audio_convert, lame);
>>>>   link_two_elements(lame, filesink);
>>>
>>> Appsink (in this case of yours) is meant to *replace* filesink. You don't
>>> seem to link appsink to anything and also you link filesink to lame. So,
>>> remove filesink from your pipeline, and use appsink in its place.
>>>
>>>>
>>>>   /* Conecting to the new-buffer signal emited by the appsink */
>>>>   g_signal_connect (appsink, "new-buffer",  G_CALLBACK (on_new_buffer),
>>>> appsink_file);
>>>>
>>>>   /* Set the pipeline to "playing" state*/
>>>>   g_print ("Now listening on port: %d\n", udp_port);
>>>>   gst_element_set_state (pipeline, GST_STATE_PLAYING);
>>>>
>>>>   /* Iterate */
>>>>   g_print ("Running...\n");
>>>>   g_main_loop_run (loop);
>>>>
>>>>   /* Out of the main loop, clean up nicely */
>>>>   g_print ("Returned, stopping listening\n");
>>>>   gst_element_set_state (pipeline, GST_STATE_NULL);
>>>>
>>>>   g_print ("Deleting pipeline\n");
>>>>   gst_object_unref (GST_OBJECT (pipeline));
>>>>   fclose(appsink_file);
>>>>
>>>>   return 0;
>>>> }
>>>>
>>>>
>>>>
>>>> best regards,
>>>> Katcipis
>>>> --
>>>> "it might be a profitable thing to learn Java, but it has no intellectual
>>>> value whatsoever" Alexander Stepanov
>>>>
>>>>
>>>> ------------------------------------------------------------------------------
>>>> Enter the BlackBerry Developer Challenge
>>>> This is your chance to win up to $100,000 in prizes! For a limited time,
>>>> vendors submitting new applications to BlackBerry App World(TM) will have
>>>> the opportunity to enter the BlackBerry Developer Challenge. See full
>>>> prize
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>>>> _______________________________________________
>>>> gstreamer-devel mailing list
>>>> [hidden email]
>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>>>>
>>>
>>>
>>>
>>> --
>>> Thiago Sousa Santos
>>>
>>> Embedded Systems and Pervasive Computing Lab (Embedded)
>>> Center of Electrical Engineering and Informatics (CEEI)
>>> Federal University of Campina Grande (UFCG)
>>>
>>>
>>> ------------------------------------------------------------------------------
>>> Enter the BlackBerry Developer Challenge
>>> This is your chance to win up to $100,000 in prizes! For a limited time,
>>> vendors submitting new applications to BlackBerry App World(TM) will have
>>> the opportunity to enter the BlackBerry Developer Challenge. See full
>>> prize
>>> details at: http://p.sf.net/sfu/Challenge
>>> _______________________________________________
>>> gstreamer-devel mailing list
>>> [hidden email]
>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>>>
>>
>>
>>
>> --
>> "it might be a profitable thing to learn Java, but it has no intellectual
>> value whatsoever" Alexander Stepanov
>>
>> ------------------------------------------------------------------------------
>> Enter the BlackBerry Developer Challenge
>> This is your chance to win up to $100,000 in prizes! For a limited time,
>> vendors submitting new applications to BlackBerry App World(TM) will have
>> the opportunity to enter the BlackBerry Developer Challenge. See full prize
>> details at: http://p.sf.net/sfu/Challenge
>> _______________________________________________
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>> [hidden email]
>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>>
>>
>
>
>
> --
> Andoni Morales Alastruey
>
> LongoMatch:The Digital Coach
> http://www.longomatch.ylatuya.es
>



--
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Re: Problem using appsink on simple example

Tiago Katcipis
In reply to this post by Andoni Morales
Thanks for the help Andoni, tomorow im going to test this.

best regards,
Katcipis

On Wed, Jul 8, 2009 at 7:51 PM, Andoni Morales <[hidden email]> wrote:
2009/7/9 Tiago Katcipis <[hidden email]>:
> Puting appsink on the place of filesink dont works too. My problem is that
> even filesink stops to work when i just add the appsink to the bin. It seens
If you need to have an element in the pipeline that it's not linked
b/c you may want to link it dinamically you need to lock it's state
usin GST_STATE_LOCK(appsink), this way state changes won't affect to
this element.
Andoni
> that only adding the appsink to the bin makes the pipe stop working.
>
> best regards,
> Katcipis
>
> On Wed, Jul 8, 2009 at 7:32 PM, thiagoss <[hidden email]> wrote:
>>
>>
>> On Wed, Jul 8, 2009 at 11:25 AM, Tiago Katcipis <[hidden email]>
>> wrote:
>>>
>>> Im going to use appsink on a project using gstreamer and started to build
>>> a basic example os]f how to use appsink using signals. My problem is that
>>> when i simply add the appsink to the pipeline, the pipe simply stops to
>>> work.
>>>
>>> On the following example im just trying to use the appsink instead the
>>> filesink to write the entire stream to a file. If i use the filesink and
>>> DONT add the appsink to the pipe, it works fine, if i just add the appsink
>>> to the pipe ....it stops working, im not even using the appsink yet, the
>>> filesink stops to write data, the result of the test will be an empty file,
>>> and no error msg is sent. I dont know what detail im missing on how to use
>>> appsink, hope someone can help me. By the way, gst_bin_add returns TRUE when
>>> i add the appsink.
>>>
>>> the source code of the example:
>>>
>>> #include <gst/gst.h>
>>> #include <glib.h>
>>> #include <gst/app/gstappsink.h>
>>> #include <stdio.h>
>>>
>>> static gboolean
>>> bus_call (GstBus     *bus,
>>>           GstMessage *msg,
>>>           gpointer    data)
>>> {
>>>   GMainLoop *loop = (GMainLoop *) data;
>>>
>>>   switch (GST_MESSAGE_TYPE (msg)) {
>>>
>>>     case GST_MESSAGE_EOS:
>>>       g_print ("End of stream\n");
>>>       g_main_loop_quit (loop);
>>>       break;
>>>
>>>     case GST_MESSAGE_ERROR: {
>>>       gchar  *debug;
>>>       GError *error;
>>>
>>>       gst_message_parse_error (msg, &error, &debug);
>>>       g_free (debug);
>>>
>>>       g_printerr ("Error: %s\n", error->message);
>>>       g_error_free (error);
>>>
>>>       g_main_loop_quit (loop);
>>>       break;
>>>     }
>>>     default:
>>>       g_print("Msg type[%d], Msg type name[%s]\n", GST_MESSAGE_TYPE(msg),
>>> GST_MESSAGE_TYPE_NAME(msg));
>>>       break;
>>>   }
>>>
>>>   return TRUE;
>>> }
>>>
>>>
>>> static void link_two_elements(GstElement* src_element, GstElement*
>>> sink_element)
>>> {
>>>   if(!gst_element_link(src_element, sink_element))
>>>       g_debug("Error linking %s to %s",
>>> gst_element_get_name(src_element), gst_element_get_name(sink_element));
>>>
>>> }
>>>
>>> static void link_two_pads(GstPad* src_pad, GstPad* sink_pad)
>>> {
>>>   if(!src_pad){
>>>       g_warning("Error: src_pad is NULL on link_two_pads");
>>>       return;
>>>   }
>>>
>>>   if(!sink_pad){
>>>       g_warning("Error: sink_pad is NULL on link_two_pads");
>>>       return;
>>>   }
>>>
>>>   if(gst_pad_link(src_pad, sink_pad) != GST_PAD_LINK_OK)
>>>       g_debug("Error linking pads %s to %s", gst_pad_get_name(src_pad),
>>> gst_pad_get_name(sink_pad));
>>>
>>> }
>>>
>>> static void on_new_buffer (GstElement* object,
>>>                            gpointer user_data)
>>> {
>>>   FILE* file = (FILE*) user_data;
>>>   GstAppSink* app_sink = (GstAppSink*) object;
>>>   GstBuffer * buffer = gst_app_sink_pull_buffer(app_sink);
>>>
>>>   if(fwrite (GST_BUFFER_DATA(buffer), 1 , GST_BUFFER_SIZE(buffer) , file)
>>> != GST_BUFFER_SIZE(buffer)){
>>>       g_debug("Error writing data from appsink to file!!!");
>>>   }else{
>>>       g_debug("Data pulled from appsink and writed to file with
>>> success!!");
>>>   }
>>>
>>> }
>>>
>>> // Pipe to test this src: gst-launch audiotestsrc ! audioconvert !
>>> alawenc ! rtppcmapay ! udpsink host=127.0.0.1 port=5000
>>> // Equivalent working pipe: gst-launch udpsrc port=5000
>>> caps=application/x-rtp ! gstrtpjitterbuffer ! rtppcmadepay ! alawdec !
>>> audioconvert ! lame ! appsink
>>> int
>>> main (int   argc,
>>>       char *argv[])
>>> {
>>>   GMainLoop *loop;
>>>
>>>   GstElement *pipeline, *source, *rtp_jitter, *rtp_alaw_depay,
>>>              *alaw_decoder, *audio_convert, *lame, *filesink, *appsink;
>>>   GstBus* bus;
>>>   GstCaps* udp_caps;
>>>   FILE* appsink_file;
>>>   int udp_port;
>>>
>>>   if(argc < 2){
>>>       g_warning("Usage: %s [port_to_be_listened]", argv[0]);
>>>       return -1;
>>>   }
>>>
>>>   udp_port = atoi(argv[1]);
>>>
>>>   /* Initialisation */
>>>   gst_init (&argc, &argv);
>>>
>>>   udp_caps = gst_caps_from_string("application/x-rtp");
>>>   if(!udp_caps){
>>>       g_warning("Error alocating the udp caps");
>>>       return -1;
>>>   }
>>>
>>>   loop = g_main_loop_new (NULL, FALSE);
>>>
>>>   /* Create gstreamer elements */
>>>   pipeline      = gst_pipeline_new("rtp-mp3-stream-decoder");
>>>   source        = gst_element_factory_make("udpsrc",  "udp-rtp-source");
>>>   rtp_jitter     = gst_element_factory_make("gstrtpjitterbuffer",
>>> "rtp-jitter-buffer");
>>>   rtp_alaw_depay = gst_element_factory_make("rtppcmadepay",
>>> "rtp_alaw_depay");
>>>   alaw_decoder   = gst_element_factory_make("alawdec","alaw-decoder");
>>>   audio_convert  =
>>> gst_element_factory_make("audioconvert","audio-convert");
>>>   lame           = gst_element_factory_make("lame","mp3-encoder");
>>>   filesink       = gst_element_factory_make("filesink",
>>> "file-mp3-output");
>>>   appsink        = gst_element_factory_make("appsink", "sink-buffer");
>>>
>>>
>>>   if (!pipeline || !source || !rtp_jitter || !appsink ||
>>>       !rtp_alaw_depay || !alaw_decoder || !audio_convert || !lame ||
>>> !filesink) {
>>>       g_printerr ("Elements could not be created. Exiting.\n");
>>>       return -1;
>>>   }
>>>
>>>   appsink_file = fopen("received_audio_appsink.mp3", "w");
>>>   if(!appsink_file){
>>>       g_printerr ("Appsink file could not be created. Exiting.\n");
>>>       return -1;
>>>   }
>>>
>>>   /* Set up the pipeline */
>>>
>>>   /* we set the properties to the source element to receive only rtp
>>> packets*/
>>>   g_object_set(G_OBJECT (source), "port", udp_port, NULL);
>>>   g_object_set(G_OBJECT (source), "caps", udp_caps, NULL);
>>>   /* we set the location of the mp3 generated file */
>>>   g_object_set(G_OBJECT (filesink), "location",
>>> "received_audio_filesink.mp3", NULL);
>>>
>>>   /*
>>>     Make appsink emit the "new-preroll" and "new-buffer" signals. This
>>> option is by default disabled because
>>>     signal emission is expensive and unneeded when the application
>>> prefers to operate in pull mode.
>>>   */
>>>   gst_app_sink_set_emit_signals ((GstAppSink*) appsink, TRUE);
>>>
>>>   /* we add a message handler */
>>>   bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
>>>   gst_bus_add_watch (bus, bus_call, loop);
>>>   gst_object_unref (bus);
>>>
>>>   /* we add all elements into the pipeline */
>>>   gst_bin_add_many (GST_BIN (pipeline),
>>>                     source, rtp_jitter, rtp_alaw_depay, alaw_decoder,
>>> audio_convert, lame, filesink, NULL);
>>>
>>>   /* When i just addn the appsink this the example stops to work and the
>>> file will be empty (and im not using the appsink, just the filesink)
>>>   if(gst_bin_add(GST_BIN (pipeline), appsink)){
>>>       g_debug("Adcionou appsink com sucesso");
>>>   }else{
>>>       g_debug("Erro ao Adcionar appsink");
>>>   }*/
>>>
>>>   /* we link all the elements together */
>>>   link_two_elements(source, rtp_jitter);
>>>   link_two_elements(rtp_jitter, rtp_alaw_depay);
>>>   link_two_elements(rtp_alaw_depay, alaw_decoder);
>>>   link_two_elements(alaw_decoder, audio_convert);
>>>   link_two_elements(audio_convert, lame);
>>>   link_two_elements(lame, filesink);
>>
>> Appsink (in this case of yours) is meant to *replace* filesink. You don't
>> seem to link appsink to anything and also you link filesink to lame. So,
>> remove filesink from your pipeline, and use appsink in its place.
>>
>>>
>>>   /* Conecting to the new-buffer signal emited by the appsink */
>>>   g_signal_connect (appsink, "new-buffer",  G_CALLBACK (on_new_buffer),
>>> appsink_file);
>>>
>>>   /* Set the pipeline to "playing" state*/
>>>   g_print ("Now listening on port: %d\n", udp_port);
>>>   gst_element_set_state (pipeline, GST_STATE_PLAYING);
>>>
>>>   /* Iterate */
>>>   g_print ("Running...\n");
>>>   g_main_loop_run (loop);
>>>
>>>   /* Out of the main loop, clean up nicely */
>>>   g_print ("Returned, stopping listening\n");
>>>   gst_element_set_state (pipeline, GST_STATE_NULL);
>>>
>>>   g_print ("Deleting pipeline\n");
>>>   gst_object_unref (GST_OBJECT (pipeline));
>>>   fclose(appsink_file);
>>>
>>>   return 0;
>>> }
>>>
>>>
>>>
>>> best regards,
>>> Katcipis
>>> --
>>> "it might be a profitable thing to learn Java, but it has no intellectual
>>> value whatsoever" Alexander Stepanov
>>>
>>>
>>> ------------------------------------------------------------------------------
>>> Enter the BlackBerry Developer Challenge
>>> This is your chance to win up to $100,000 in prizes! For a limited time,
>>> vendors submitting new applications to BlackBerry App World(TM) will have
>>> the opportunity to enter the BlackBerry Developer Challenge. See full
>>> prize
>>> details at: http://p.sf.net/sfu/Challenge
>>> _______________________________________________
>>> gstreamer-devel mailing list
>>> [hidden email]
>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>>>
>>
>>
>>
>> --
>> Thiago Sousa Santos
>>
>> Embedded Systems and Pervasive Computing Lab (Embedded)
>> Center of Electrical Engineering and Informatics (CEEI)
>> Federal University of Campina Grande (UFCG)
>>
>>
>> ------------------------------------------------------------------------------
>> Enter the BlackBerry Developer Challenge
>> This is your chance to win up to $100,000 in prizes! For a limited time,
>> vendors submitting new applications to BlackBerry App World(TM) will have
>> the opportunity to enter the BlackBerry Developer Challenge. See full
>> prize
>> details at: http://p.sf.net/sfu/Challenge
>> _______________________________________________
>> gstreamer-devel mailing list
>> [hidden email]
>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>>
>
>
>
> --
> "it might be a profitable thing to learn Java, but it has no intellectual
> value whatsoever" Alexander Stepanov
>
> ------------------------------------------------------------------------------
> Enter the BlackBerry Developer Challenge
> This is your chance to win up to $100,000 in prizes! For a limited time,
> vendors submitting new applications to BlackBerry App World(TM) will have
> the opportunity to enter the BlackBerry Developer Challenge. See full prize
> details at: http://p.sf.net/sfu/Challenge
> _______________________________________________
> gstreamer-devel mailing list
> [hidden email]
> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>
>



--
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http://www.longomatch.ylatuya.es

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Re: Problem using appsink on simple example

Tiago Katcipis
In reply to this post by Tim-Philipp Müller-2
Thanks for the help Tim, im new at gstreamer, i already made some aplications but nothing too advanced, now i have to do more sofisticated things involving appsink and appsrc.

best regards,
Katcipis

On Wed, Jul 8, 2009 at 1:11 PM, Tim-Philipp Müller <[hidden email]> wrote:
On Wed, 2009-07-08 at 11:25 -0300, Tiago Katcipis wrote:

> If i use the filesink and DONT add the appsink to the pipe, it works
> fine. If i just add the appsink to the pipe ....it stops working, im
> not even using the appsink yet, the filesink stops to write data, the
> result of the test will be an empty file, and no error msg is sent.

This is somewhat expected. The pipeline is unlikely to preroll (go to
>=PAUSED state) if there's an unconnected sink element in it, since it
will wait for that sink to receive data, but if it's not connected the
sink will never receive any data.

Cheers
 -Tim



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Re: Problem using appsink on simple example

Tiago Katcipis
It may help another newbies like me :-), i got the my alaw to mp3 transcoder using appsink to write a file working, to someone starting on gstreamer it may be usefull.

https://svn.inf.ufsc.br/katcipis/c/gstreamer/receiver_rtp_alaw_to_mp3_appsink_signal

on firefox it may give some warnings about the certificate, its because firefox dont recognize UFSC certificate.

best regards and thanks again for the help,
Katcipis

On Thu, Jul 9, 2009 at 4:19 PM, Tiago Katcipis <[hidden email]> wrote:
Thanks for the help Tim, im new at gstreamer, i already made some aplications but nothing too advanced, now i have to do more sofisticated things involving appsink and appsrc.

best regards,
Katcipis


On Wed, Jul 8, 2009 at 1:11 PM, Tim-Philipp Müller <[hidden email]> wrote:
On Wed, 2009-07-08 at 11:25 -0300, Tiago Katcipis wrote:

> If i use the filesink and DONT add the appsink to the pipe, it works
> fine. If i just add the appsink to the pipe ....it stops working, im
> not even using the appsink yet, the filesink stops to write data, the
> result of the test will be an empty file, and no error msg is sent.

This is somewhat expected. The pipeline is unlikely to preroll (go to
>=PAUSED state) if there's an unconnected sink element in it, since it
will wait for that sink to receive data, but if it's not connected the
sink will never receive any data.

Cheers
 -Tim



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Re: Problem using appsink on simple example

luca101
Hello Tiago,

Do you still have the appsink code that you posted at:
https://svn.inf.ufsc.br/katcipis/c/gstreamer/receiver_rtp_alaw_to_mp3_appsink_signal

The link above is broken and I have the same problem with appsink.

Thanks,

Luca
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Re: Problem using appsink on simple example

darksaint
In reply to this post by Tiago Katcipis
I too am starting to dive into AppSink and would love to view your sample code that you got working.

Thanks.
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