gstreamer/tests/examples/app/appsink-src2.c
2015-04-07 15:17:46 +01:00

199 lines
5.8 KiB
C

/* GStreamer
*
* appsrc-src2.c: example for using gst_app_src_push_sample().
*
* Copyright (C) 2014 Nicola Murino <nicola.murino@gmail.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 <gst/gst.h>
#include <string.h>
#include <gst/app/gstappsrc.h>
#include <gst/app/gstappsink.h>
/*
* In this sample we show the usage of gst_app_src_push_sample in push
* mode, this method set the appsrc caps based on the caps from the sample
*
*/
typedef struct
{
GMainLoop *loop;
GstElement *source;
GstElement *sink;
} ProgramData;
/* called when the appsink notifies us that there is a new buffer ready for
* processing */
static GstFlowReturn
on_new_sample_from_sink (GstElement * elt, ProgramData * data)
{
GstSample *sample;
GstElement *source;
GstFlowReturn ret;
/* get the sample from appsink */
sample = gst_app_sink_pull_sample (GST_APP_SINK (elt));
/* get source an push new sample */
source = gst_bin_get_by_name (GST_BIN (data->sink), "testsource");
ret = gst_app_src_push_sample (GST_APP_SRC (source), sample);
gst_object_unref (source);
/* we don't need the appsink sample anymore */
gst_sample_unref (sample);
return ret;
}
/* called when we get a GstMessage from the source pipeline when we get EOS, we
* notify the appsrc of it. */
static gboolean
on_source_message (GstBus * bus, GstMessage * message, ProgramData * data)
{
GstElement *source;
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_EOS:
g_print ("The source got dry\n");
source = gst_bin_get_by_name (GST_BIN (data->sink), "testsource");
gst_app_src_end_of_stream (GST_APP_SRC (source));
gst_object_unref (source);
break;
case GST_MESSAGE_ERROR:
g_print ("Received error\n");
g_main_loop_quit (data->loop);
break;
default:
break;
}
return TRUE;
}
/* called when we get a GstMessage from the sink pipeline when we get EOS, we
* exit the mainloop and this testapp. */
static gboolean
on_sink_message (GstBus * bus, GstMessage * message, ProgramData * data)
{
/* nil */
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_EOS:
g_print ("Finished playback\n");
g_main_loop_quit (data->loop);
break;
case GST_MESSAGE_ERROR:
g_print ("Received error\n");
g_main_loop_quit (data->loop);
break;
default:
break;
}
return TRUE;
}
int
main (int argc, char *argv[])
{
gchar *filename = NULL;
ProgramData *data = NULL;
gchar *string = NULL;
GstBus *bus = NULL;
GstElement *testsink = NULL;
GstElement *testsource = NULL;
gst_init (&argc, &argv);
data = g_new0 (ProgramData, 1);
data->loop = g_main_loop_new (NULL, FALSE);
/* setting up source pipeline, we read from a file and convert to our desired
* caps. */
string =
g_strdup_printf
("audiotestsrc num-buffers=200 ! wavenc ! wavparse ! appsink name=testsink");
g_free (filename);
data->source = gst_parse_launch (string, NULL);
g_free (string);
if (data->source == NULL) {
g_print ("Bad source\n");
return -1;
}
/* to be notified of messages from this pipeline, mostly EOS */
bus = gst_element_get_bus (data->source);
gst_bus_add_watch (bus, (GstBusFunc) on_source_message, data);
gst_object_unref (bus);
/* we use appsink in push mode, it sends us a signal when data is available
* and we pull out the data in the signal callback. We want the appsink to
* push as fast as it can, hence the sync=false */
testsink = gst_bin_get_by_name (GST_BIN (data->source), "testsink");
g_object_set (G_OBJECT (testsink), "emit-signals", TRUE, "sync", FALSE, NULL);
g_signal_connect (testsink, "new-sample",
G_CALLBACK (on_new_sample_from_sink), data);
gst_object_unref (testsink);
/* setting up sink pipeline, we push audio data into this pipeline that will
* then play it back using the default audio sink. */
string =
g_strdup_printf
("appsrc name=testsource ! audioconvert ! audioresample ! autoaudiosink");
data->sink = gst_parse_launch (string, NULL);
g_free (string);
if (data->sink == NULL) {
g_print ("Bad sink\n");
return -1;
}
testsource = gst_bin_get_by_name (GST_BIN (data->sink), "testsource");
/* configure for time-based format */
g_object_set (testsource, "format", GST_FORMAT_TIME, NULL);
/* uncomment the next line to block when appsrc has buffered enough */
/* g_object_set (testsource, "block", TRUE, NULL); */
gst_object_unref (testsource);
bus = gst_element_get_bus (data->sink);
gst_bus_add_watch (bus, (GstBusFunc) on_sink_message, data);
gst_object_unref (bus);
/* launching things */
gst_element_set_state (data->sink, GST_STATE_PLAYING);
gst_element_set_state (data->source, GST_STATE_PLAYING);
/* let's run !, this loop will quit when the sink pipeline goes EOS or when an
* error occurs in the source or sink pipelines. */
g_print ("Let's run!\n");
g_main_loop_run (data->loop);
g_print ("Going out\n");
gst_element_set_state (data->source, GST_STATE_NULL);
gst_element_set_state (data->sink, GST_STATE_NULL);
gst_object_unref (data->source);
gst_object_unref (data->sink);
g_main_loop_unref (data->loop);
g_free (data);
return 0;
}