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201 lines
5.9 KiB
C
201 lines
5.9 KiB
C
/* GStreamer
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*
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* appsrc-src2.c: example for using gst_app_src_push_sample().
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*
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* Copyright (C) 2014 Nicola Murino <nicola.murino@gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#include <gst/gst.h>
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#include <string.h>
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#include <gst/app/gstappsrc.h>
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#include <gst/app/gstappsink.h>
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/*
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* In this sample we show the usage of gst_app_src_push_sample in push
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* mode, this method set the appsrc caps based on the caps from the sample
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*
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*/
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typedef struct
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{
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GMainLoop *loop;
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GstElement *source;
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GstElement *sink;
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} ProgramData;
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/* called when the appsink notifies us that there is a new buffer ready for
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* processing */
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static GstFlowReturn
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on_new_sample_from_sink (GstElement * elt, ProgramData * data)
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{
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GstSample *sample;
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GstElement *source;
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GstFlowReturn ret;
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/* get the sample from appsink */
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sample = gst_app_sink_pull_sample (GST_APP_SINK (elt));
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/* get source an push new sample */
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source = gst_bin_get_by_name (GST_BIN (data->sink), "testsource");
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ret = gst_app_src_push_sample (GST_APP_SRC (source), sample);
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gst_object_unref (source);
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/* we don't need the appsink sample anymore */
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gst_sample_unref (sample);
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return ret;
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}
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/* called when we get a GstMessage from the source pipeline when we get EOS, we
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* notify the appsrc of it. */
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static gboolean
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on_source_message (GstBus * bus, GstMessage * message, ProgramData * data)
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{
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GstElement *source;
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switch (GST_MESSAGE_TYPE (message)) {
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case GST_MESSAGE_EOS:
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g_print ("The source got dry\n");
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source = gst_bin_get_by_name (GST_BIN (data->sink), "testsource");
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gst_app_src_end_of_stream (GST_APP_SRC (source));
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gst_object_unref (source);
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break;
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case GST_MESSAGE_ERROR:
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g_print ("Received error\n");
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g_main_loop_quit (data->loop);
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break;
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default:
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break;
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}
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return TRUE;
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}
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/* called when we get a GstMessage from the sink pipeline when we get EOS, we
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* exit the mainloop and this testapp. */
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static gboolean
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on_sink_message (GstBus * bus, GstMessage * message, ProgramData * data)
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{
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/* nil */
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switch (GST_MESSAGE_TYPE (message)) {
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case GST_MESSAGE_EOS:
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g_print ("Finished playback\n");
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g_main_loop_quit (data->loop);
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break;
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case GST_MESSAGE_ERROR:
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g_print ("Received error\n");
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g_main_loop_quit (data->loop);
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break;
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default:
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break;
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}
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return TRUE;
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}
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int
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main (int argc, char *argv[])
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{
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ProgramData *data = NULL;
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gchar *string = NULL;
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GstBus *bus = NULL;
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GstElement *testsink = NULL;
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GstElement *testsource = NULL;
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gst_init (&argc, &argv);
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data = g_new0 (ProgramData, 1);
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data->loop = g_main_loop_new (NULL, FALSE);
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/* setting up source pipeline, we read from a file and convert to our desired
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* caps. */
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string =
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g_strdup_printf
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("audiotestsrc num-buffers=200 ! wavenc ! wavparse ! appsink name=testsink");
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data->source = gst_parse_launch (string, NULL);
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g_free (string);
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if (data->source == NULL) {
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g_print ("Bad source\n");
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g_main_loop_unref (data->loop);
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g_free (data);
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return -1;
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}
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/* to be notified of messages from this pipeline, mostly EOS */
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bus = gst_element_get_bus (data->source);
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gst_bus_add_watch (bus, (GstBusFunc) on_source_message, data);
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gst_object_unref (bus);
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/* we use appsink in push mode, it sends us a signal when data is available
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* and we pull out the data in the signal callback. We want the appsink to
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* push as fast as it can, hence the sync=false */
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testsink = gst_bin_get_by_name (GST_BIN (data->source), "testsink");
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g_object_set (G_OBJECT (testsink), "emit-signals", TRUE, "sync", FALSE, NULL);
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g_signal_connect (testsink, "new-sample",
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G_CALLBACK (on_new_sample_from_sink), data);
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gst_object_unref (testsink);
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/* setting up sink pipeline, we push audio data into this pipeline that will
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* then play it back using the default audio sink. */
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string =
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g_strdup_printf
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("appsrc name=testsource ! audioconvert ! audioresample ! autoaudiosink");
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data->sink = gst_parse_launch (string, NULL);
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g_free (string);
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if (data->sink == NULL) {
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g_print ("Bad sink\n");
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g_main_loop_unref (data->loop);
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g_free (data);
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return -1;
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}
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testsource = gst_bin_get_by_name (GST_BIN (data->sink), "testsource");
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/* configure for time-based format */
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g_object_set (testsource, "format", GST_FORMAT_TIME, NULL);
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/* uncomment the next line to block when appsrc has buffered enough */
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/* g_object_set (testsource, "block", TRUE, NULL); */
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gst_object_unref (testsource);
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bus = gst_element_get_bus (data->sink);
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gst_bus_add_watch (bus, (GstBusFunc) on_sink_message, data);
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gst_object_unref (bus);
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/* launching things */
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gst_element_set_state (data->sink, GST_STATE_PLAYING);
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gst_element_set_state (data->source, GST_STATE_PLAYING);
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/* let's run !, this loop will quit when the sink pipeline goes EOS or when an
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* error occurs in the source or sink pipelines. */
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g_print ("Let's run!\n");
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g_main_loop_run (data->loop);
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g_print ("Going out\n");
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gst_element_set_state (data->source, GST_STATE_NULL);
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gst_element_set_state (data->sink, GST_STATE_NULL);
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gst_object_unref (data->source);
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gst_object_unref (data->sink);
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g_main_loop_unref (data->loop);
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g_free (data);
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return 0;
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}
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