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#include <gst/gst.h>
#include <string.h>
#include <gst/app/gstappsrc.h>
#include <gst/app/gstappsink.h>
#include <gst/app/gstappbuffer.h>
/* these are the caps we are going to pass through the appsink and appsrc */
const gchar *audio_caps =
"audio/x-raw-int,channels=1,rate=8000,signed=(boolean)true,width=16,depth=16,endianness=1234";
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 void
on_new_buffer_from_source (GstElement * elt, ProgramData * data)
{
guint size;
gpointer raw_buffer;
GstBuffer *app_buffer, *buffer;
GstElement *source;
/* get the buffer from appsink */
buffer = gst_app_sink_pull_buffer (GST_APP_SINK (elt));
/* turn it into an app buffer, it's not really needed, we could simply push
* the retrieved buffer from appsink into appsrc just fine. */
size = GST_BUFFER_SIZE (buffer);
g_print ("Pushing a buffer of size %d\n", size);
raw_buffer = g_malloc0 (size);
memcpy (raw_buffer, GST_BUFFER_DATA (buffer), size);
app_buffer = gst_app_buffer_new (raw_buffer, size, g_free, raw_buffer);
/* newer basesrc will set caps for use automatically but it does not really
* hurt to set it on the buffer again */
gst_buffer_set_caps (app_buffer, GST_BUFFER_CAPS (buffer));
/* we don't need the appsink buffer anymore */
gst_buffer_unref (buffer);
/* get source an push new buffer */
source = gst_bin_get_by_name (GST_BIN (data->sink), "testsource");
gst_app_src_push_buffer (GST_APP_SRC (source), app_buffer);
}
/* 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));
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);
if (argc == 2)
filename = g_strdup (argv[1]);
else
filename = g_strdup ("/usr/share/sounds/ekiga/ring.wav");
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
("filesrc location=\"%s\" ! wavparse ! audioconvert ! audioresample ! appsink caps=\"%s\" name=testsink",
filename, audio_caps);
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-buffer",
G_CALLBACK (on_new_buffer_from_source), 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. We have no blocking
* behaviour on the src which means that we will push the entire file into
* memory. */
string =
g_strdup_printf ("appsrc name=testsource caps=\"%s\" ! autoaudiosink",
audio_caps);
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;
}
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