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authorSebastian Dröge <sebastian.droege@collabora.co.uk>2009-02-24 16:33:51 +0100
committerSebastian Dröge <sebastian.droege@collabora.co.uk>2009-02-24 16:33:51 +0100
commitf0ae68d9440e90711c7b547f03d7165357b9c4ac (patch)
treed77797c994b5f7457b1dd3b802f08f783c8ce0e5 /tests/examples/app/appsrc-stream.c
parentda040c2a3529801e9b7c239afb698379ede89087 (diff)
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Move examples directory to tests/examples as in every other GStreamer module
Diffstat (limited to 'tests/examples/app/appsrc-stream.c')
-rw-r--r--tests/examples/app/appsrc-stream.c249
1 files changed, 249 insertions, 0 deletions
diff --git a/tests/examples/app/appsrc-stream.c b/tests/examples/app/appsrc-stream.c
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+/* GStreamer
+ *
+ * appsrc-stream.c: example for using appsrc in streaming mode.
+ *
+ * Copyright (C) 2008 Wim Taymans <wim.taymans@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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <gst/gst.h>
+
+#include <stdio.h>
+#include <string.h>
+#include <stdlib.h>
+
+GST_DEBUG_CATEGORY (appsrc_playbin_debug);
+#define GST_CAT_DEFAULT appsrc_playbin_debug
+
+/*
+ * an example application of using appsrc in streaming push mode. We simply push
+ * buffers into appsrc. The size of the buffers we push can be any size we
+ * choose.
+ *
+ * This example is very close to how one would deal with a streaming webserver
+ * that does not support range requests or does not report the total file size.
+ *
+ * Some optimisations are done so that we don't push too much data. We connect
+ * to the need-data and enough-data signals to start/stop sending buffers.
+ *
+ * Appsrc in streaming mode (the default) does not support seeking so we don't
+ * have to handle any seek callbacks.
+ *
+ * Some formats are able to estimate the duration of the media file based on the
+ * file length (mp3, mpeg,..), others report an unknown length (ogg,..).
+ */
+typedef struct _App App;
+
+struct _App
+{
+ GstElement *playbin;
+ GstElement *appsrc;
+
+ GMainLoop *loop;
+ guint sourceid;
+
+ GMappedFile *file;
+ guint8 *data;
+ gsize length;
+ guint64 offset;
+};
+
+App s_app;
+
+#define CHUNK_SIZE 4096
+
+/* This method is called by the idle GSource in the mainloop. We feed CHUNK_SIZE
+ * bytes into appsrc.
+ * The ide handler is added to the mainloop when appsrc requests us to start
+ * sending data (need-data signal) and is removed when appsrc has enough data
+ * (enough-data signal).
+ */
+static gboolean
+read_data (App * app)
+{
+ GstBuffer *buffer;
+ guint len;
+ GstFlowReturn ret;
+
+ buffer = gst_buffer_new ();
+
+ if (app->offset >= app->length) {
+ /* we are EOS, send end-of-stream and remove the source */
+ g_signal_emit_by_name (app->appsrc, "end-of-stream", &ret);
+ return FALSE;
+ }
+
+ /* read the next chunk */
+ len = CHUNK_SIZE;
+ if (app->offset + len > app->length)
+ len = app->length - app->offset;
+
+ GST_BUFFER_DATA (buffer) = app->data + app->offset;
+ GST_BUFFER_SIZE (buffer) = len;
+
+ GST_DEBUG ("feed buffer %p, offset %" G_GUINT64_FORMAT "-%u", buffer,
+ app->offset, len);
+ g_signal_emit_by_name (app->appsrc, "push-buffer", buffer, &ret);
+ gst_buffer_unref (buffer);
+ if (ret != GST_FLOW_OK) {
+ /* some error, stop sending data */
+ return FALSE;
+ }
+
+ app->offset += len;
+
+ return TRUE;
+}
+
+/* This signal callback is called when appsrc needs data, we add an idle handler
+ * to the mainloop to start pushing data into the appsrc */
+static void
+start_feed (GstElement * playbin, guint size, App * app)
+{
+ if (app->sourceid == 0) {
+ GST_DEBUG ("start feeding");
+ app->sourceid = g_idle_add ((GSourceFunc) read_data, app);
+ }
+}
+
+/* This callback is called when appsrc has enough data and we can stop sending.
+ * We remove the idle handler from the mainloop */
+static void
+stop_feed (GstElement * playbin, App * app)
+{
+ if (app->sourceid != 0) {
+ GST_DEBUG ("stop feeding");
+ g_source_remove (app->sourceid);
+ app->sourceid = 0;
+ }
+}
+
+/* this callback is called when playbin2 has constructed a source object to read
+ * from. Since we provided the appsrc:// uri to playbin2, this will be the
+ * appsrc that we must handle. We set up some signals to start and stop pushing
+ * data into appsrc */
+static void
+found_source (GObject * object, GObject * orig, GParamSpec * pspec, App * app)
+{
+ /* get a handle to the appsrc */
+ g_object_get (orig, pspec->name, &app->appsrc, NULL);
+
+ GST_DEBUG ("got appsrc %p", app->appsrc);
+
+ /* we can set the length in appsrc. This allows some elements to estimate the
+ * total duration of the stream. It's a good idea to set the property when you
+ * can but it's not required. */
+ g_object_set (app->appsrc, "size", app->length, NULL);
+
+ /* configure the appsrc, we will push data into the appsrc from the
+ * mainloop. */
+ g_signal_connect (app->appsrc, "need-data", G_CALLBACK (start_feed), app);
+ g_signal_connect (app->appsrc, "enough-data", G_CALLBACK (stop_feed), app);
+}
+
+static gboolean
+bus_message (GstBus * bus, GstMessage * message, App * app)
+{
+ GST_DEBUG ("got message %s",
+ gst_message_type_get_name (GST_MESSAGE_TYPE (message)));
+
+ switch (GST_MESSAGE_TYPE (message)) {
+ case GST_MESSAGE_ERROR:
+ g_error ("received error");
+ g_main_loop_quit (app->loop);
+ break;
+ case GST_MESSAGE_EOS:
+ g_main_loop_quit (app->loop);
+ break;
+ default:
+ break;
+ }
+ return TRUE;
+}
+
+int
+main (int argc, char *argv[])
+{
+ App *app = &s_app;
+ GError *error = NULL;
+ GstBus *bus;
+
+ gst_init (&argc, &argv);
+
+ GST_DEBUG_CATEGORY_INIT (appsrc_playbin_debug, "appsrc-playbin", 0,
+ "appsrc playbin example");
+
+ if (argc < 2) {
+ g_print ("usage: %s <filename>\n", argv[0]);
+ return -1;
+ }
+
+ /* try to open the file as an mmapped file */
+ app->file = g_mapped_file_new (argv[1], FALSE, &error);
+ if (error) {
+ g_print ("failed to open file: %s\n", error->message);
+ g_error_free (error);
+ return -2;
+ }
+ /* get some vitals, this will be used to read data from the mmapped file and
+ * feed it to appsrc. */
+ app->length = g_mapped_file_get_length (app->file);
+ app->data = (guint8 *) g_mapped_file_get_contents (app->file);
+ app->offset = 0;
+
+ /* create a mainloop to get messages and to handle the idle handler that will
+ * feed data to appsrc. */
+ app->loop = g_main_loop_new (NULL, TRUE);
+
+ app->playbin = gst_element_factory_make ("playbin2", NULL);
+ g_assert (app->playbin);
+
+ bus = gst_pipeline_get_bus (GST_PIPELINE (app->playbin));
+
+ /* add watch for messages */
+ gst_bus_add_watch (bus, (GstBusFunc) bus_message, app);
+
+ /* set to read from appsrc */
+ g_object_set (app->playbin, "uri", "appsrc://", NULL);
+
+ /* get notification when the source is created so that we get a handle to it
+ * and can configure it */
+ g_signal_connect (app->playbin, "deep-notify::source",
+ (GCallback) found_source, app);
+
+ /* go to playing and wait in a mainloop. */
+ gst_element_set_state (app->playbin, GST_STATE_PLAYING);
+
+ /* this mainloop is stopped when we receive an error or EOS */
+ g_main_loop_run (app->loop);
+
+ GST_DEBUG ("stopping");
+
+ gst_element_set_state (app->playbin, GST_STATE_NULL);
+
+ /* free the file */
+ g_mapped_file_free (app->file);
+
+ gst_object_unref (bus);
+ g_main_loop_unref (app->loop);
+
+ return 0;
+}