gstreamer/subprojects/gst-docs/markdown/tutorials/playback/short-cutting-the-pipeline.md

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# Playback tutorial 3: Short-cutting the pipeline
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{{ ALERT_PY.md }}
{{ ALERT_JS.md }}
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## Goal
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[](tutorials/basic/short-cutting-the-pipeline.md) showed
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how an application can manually extract or inject data into a pipeline
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by using two special elements called `appsrc` and `appsink`.
`playbin` allows using these elements too, but the method to connect
them is different. To connect an `appsink` to `playbin` see [](tutorials/playback/custom-playbin-sinks.md).
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This tutorial shows:
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- How to connect `appsrc` with `playbin`
- How to configure the `appsrc`
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## A playbin waveform generator
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Copy this code into a text file named `playback-tutorial-3.c`.
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**playback-tutorial-3.c**
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``` c
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#include <gst/gst.h>
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#include <gst/audio/audio.h>
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#include <string.h>
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#define CHUNK_SIZE 1024 /* Amount of bytes we are sending in each buffer */
#define SAMPLE_RATE 44100 /* Samples per second we are sending */
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/* Structure to contain all our information, so we can pass it to callbacks */
typedef struct _CustomData {
GstElement *pipeline;
GstElement *app_source;
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guint64 num_samples; /* Number of samples generated so far (for timestamp generation) */
gfloat a, b, c, d; /* For waveform generation */
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guint sourceid; /* To control the GSource */
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GMainLoop *main_loop; /* GLib's Main Loop */
} CustomData;
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/* This method is called by the idle GSource in the mainloop, to 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 push_data (CustomData *data) {
GstBuffer *buffer;
GstFlowReturn ret;
int i;
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GstMapInfo map;
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gint16 *raw;
gint num_samples = CHUNK_SIZE / 2; /* Because each sample is 16 bits */
gfloat freq;
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/* Create a new empty buffer */
buffer = gst_buffer_new_and_alloc (CHUNK_SIZE);
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/* Set its timestamp and duration */
GST_BUFFER_TIMESTAMP (buffer) = gst_util_uint64_scale (data->num_samples, GST_SECOND, SAMPLE_RATE);
GST_BUFFER_DURATION (buffer) = gst_util_uint64_scale (num_samples, GST_SECOND, SAMPLE_RATE);
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/* Generate some psychodelic waveforms */
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gst_buffer_map (buffer, &map, GST_MAP_WRITE);
raw = (gint16 *)map.data;
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data->c += data->d;
data->d -= data->c / 1000;
freq = 1100 + 1000 * data->d;
for (i = 0; i < num_samples; i++) {
data->a += data->b;
data->b -= data->a / freq;
raw[i] = (gint16)(500 * data->a);
}
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gst_buffer_unmap (buffer, &map);
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data->num_samples += num_samples;
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/* Push the buffer into the appsrc */
g_signal_emit_by_name (data->app_source, "push-buffer", buffer, &ret);
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/* Free the buffer now that we are done with it */
gst_buffer_unref (buffer);
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if (ret != GST_FLOW_OK) {
/* We got some error, stop sending data */
return FALSE;
}
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return TRUE;
}
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/* This signal callback triggers when appsrc needs data. Here, we add an idle handler
* to the mainloop to start pushing data into the appsrc */
static void start_feed (GstElement *source, guint size, CustomData *data) {
if (data->sourceid == 0) {
g_print ("Start feeding\n");
data->sourceid = g_idle_add ((GSourceFunc) push_data, data);
}
}
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/* This callback triggers when appsrc has enough data and we can stop sending.
* We remove the idle handler from the mainloop */
static void stop_feed (GstElement *source, CustomData *data) {
if (data->sourceid != 0) {
g_print ("Stop feeding\n");
g_source_remove (data->sourceid);
data->sourceid = 0;
}
}
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/* This function is called when an error message is posted on the bus */
static void error_cb (GstBus *bus, GstMessage *msg, CustomData *data) {
GError *err;
gchar *debug_info;
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/* Print error details on the screen */
gst_message_parse_error (msg, &err, &debug_info);
g_printerr ("Error received from element %s: %s\n", GST_OBJECT_NAME (msg->src), err->message);
g_printerr ("Debugging information: %s\n", debug_info ? debug_info : "none");
g_clear_error (&err);
g_free (debug_info);
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g_main_loop_quit (data->main_loop);
}
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/* This function is called when playbin has created the appsrc element, so we have
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* a chance to configure it. */
static void source_setup (GstElement *pipeline, GstElement *source, CustomData *data) {
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GstAudioInfo info;
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GstCaps *audio_caps;
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g_print ("Source has been created. Configuring.\n");
data->app_source = source;
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/* Configure appsrc */
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gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16, SAMPLE_RATE, 1, NULL);
audio_caps = gst_audio_info_to_caps (&info);
g_object_set (source, "caps", audio_caps, "format", GST_FORMAT_TIME, NULL);
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g_signal_connect (source, "need-data", G_CALLBACK (start_feed), data);
g_signal_connect (source, "enough-data", G_CALLBACK (stop_feed), data);
gst_caps_unref (audio_caps);
}
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int main(int argc, char *argv[]) {
CustomData data;
GstBus *bus;
/* Initialize custom data structure */
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memset (&data, 0, sizeof (data));
data.b = 1; /* For waveform generation */
data.d = 1;
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/* Initialize GStreamer */
gst_init (&argc, &argv);
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/* Create the playbin element */
data.pipeline = gst_parse_launch ("playbin uri=appsrc://", NULL);
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g_signal_connect (data.pipeline, "source-setup", G_CALLBACK (source_setup), &data);
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/* Instruct the bus to emit signals for each received message, and connect to the interesting signals */
bus = gst_element_get_bus (data.pipeline);
gst_bus_add_signal_watch (bus);
g_signal_connect (G_OBJECT (bus), "message::error", (GCallback)error_cb, &data);
gst_object_unref (bus);
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/* Start playing the pipeline */
gst_element_set_state (data.pipeline, GST_STATE_PLAYING);
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/* Create a GLib Main Loop and set it to run */
data.main_loop = g_main_loop_new (NULL, FALSE);
g_main_loop_run (data.main_loop);
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/* Free resources */
gst_element_set_state (data.pipeline, GST_STATE_NULL);
gst_object_unref (data.pipeline);
return 0;
}
```
> ![information] If you need help to compile this code, refer to the
> **Building the tutorials** section for your platform: [Mac] or
> [Windows] or use this specific command on Linux:
>
> `` gcc playback-tutorial-3.c -o playback-tutorial-3 `pkg-config --cflags --libs gstreamer-1.0 gstreamer-audio-1.0` ``
>
> If you need help to run this code, refer to the **Running the
> tutorials** section for your platform: [Mac OS X], [Windows][1], for
> [iOS] or for [android].
>
> This tutorial opens a window and displays a movie, with accompanying
> audio. The media is fetched from the Internet, so the window might
> take a few seconds to appear, depending on your connection
> speed. In the console window, you should see a message indicating
> where the media is being stored, and a text graph representing the
> downloaded portions and the current position. A buffering message
> appears whenever buffering is required, which might never happen is
> your network connection is fast enough
>
> Required libraries: `gstreamer-1.0` `gstreamer-audio-1.0`
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To use an `appsrc` as the source for the pipeline, simply instantiate a
`playbin` and set its URI to `appsrc://`
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``` c
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/* Create the playbin element */
data.pipeline = gst_parse_launch ("playbin uri=appsrc://", NULL);
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```
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`playbin` will create an internal `appsrc` element and fire the
`source-setup` signal to allow the application to configure
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it:
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``` c
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g_signal_connect (data.pipeline, "source-setup", G_CALLBACK (source_setup), &data);
```
In particular, it is important to set the caps property of `appsrc`,
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since, once the signal handler returns, `playbin` will instantiate the
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next element in the pipeline according to these
caps:
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``` c
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/* This function is called when playbin has created the appsrc element, so we have
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* a chance to configure it. */
static void source_setup (GstElement *pipeline, GstElement *source, CustomData *data) {
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GstAudioInfo info;
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GstCaps *audio_caps;
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g_print ("Source has been created. Configuring.\n");
data->app_source = source;
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/* Configure appsrc */
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gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16, SAMPLE_RATE, 1, NULL);
audio_caps = gst_audio_info_to_caps (&info);
g_object_set (source, "caps", audio_caps, "format", GST_FORMAT_TIME, NULL);
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g_signal_connect (source, "need-data", G_CALLBACK (start_feed), data);
g_signal_connect (source, "enough-data", G_CALLBACK (stop_feed), data);
gst_caps_unref (audio_caps);
}
```
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The configuration of the `appsrc` is exactly the same as in
[](tutorials/basic/short-cutting-the-pipeline.md):
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the caps are set to `audio/x-raw`, and two callbacks are registered,
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so the element can tell the application when it needs to start and stop
pushing data. See [](tutorials/basic/short-cutting-the-pipeline.md)
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for more details.
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From this point onwards, `playbin` takes care of the rest of the
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pipeline, and the application only needs to worry about generating more
data when told so.
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To learn how data can be extracted from `playbin` using the
`appsink` element, see [](tutorials/playback/custom-playbin-sinks.md).
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## Conclusion
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This tutorial applies the concepts shown in
[](tutorials/basic/short-cutting-the-pipeline.md) to
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`playbin`. In particular, it has shown:
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- How to connect `appsrc` with `playbin` using the special
URI `appsrc://`
- How to configure the `appsrc` using the `source-setup` signal
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It has been a pleasure having you here, and see you soon!