[gst-devel] Problem using appsink on simple example
thiagoss
thiagossantos at gmail.com
Thu Jul 9 00:32:46 CEST 2009
On Wed, Jul 8, 2009 at 11:25 AM, Tiago Katcipis <katcipis at inf.ufsc.br>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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