# Playback tutorial 3: Short-cutting the pipeline ## Goal [](sdk-basic-tutorial-short-cutting-the-pipeline.md) showed how an application can manually extract or inject data into a pipeline 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 [](sdk-playback-tutorial-custom-playbin-sinks.md). This tutorial shows: - How to connect `appsrc` with `playbin` - How to configure the `appsrc` ## A playbin waveform generator Copy this code into a text file named `playback-tutorial-3.c`. **playback-tutorial-3.c** ``` c #include #include #include #define CHUNK_SIZE 1024 /* Amount of bytes we are sending in each buffer */ #define SAMPLE_RATE 44100 /* Samples per second we are sending */ /* Structure to contain all our information, so we can pass it to callbacks */ typedef struct _CustomData { GstElement *pipeline; GstElement *app_source; guint64 num_samples; /* Number of samples generated so far (for timestamp generation) */ gfloat a, b, c, d; /* For waveform generation */ guint sourceid; /* To control the GSource */ GMainLoop *main_loop; /* GLib's Main Loop */ } CustomData; /* 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; GstMapInfo map; gint16 *raw; gint num_samples = CHUNK_SIZE / 2; /* Because each sample is 16 bits */ gfloat freq; /* Create a new empty buffer */ buffer = gst_buffer_new_and_alloc (CHUNK_SIZE); /* 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 (CHUNK_SIZE, GST_SECOND, SAMPLE_RATE); /* Generate some psychodelic waveforms */ gst_buffer_map (buffer, &map, GST_MAP_WRITE); raw = (gint16 *)map.data; 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); } gst_buffer_unmap (buffer, &map); data->num_samples += num_samples; /* Push the buffer into the appsrc */ g_signal_emit_by_name (data->app_source, "push-buffer", buffer, &ret); /* Free the buffer now that we are done with it */ gst_buffer_unref (buffer); if (ret != GST_FLOW_OK) { /* We got some error, stop sending data */ return FALSE; } return TRUE; } /* 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); } } /* 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; } } /* 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; /* 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); g_main_loop_quit (data->main_loop); } /* This function is called when playbin has created the appsrc element, so we have * a chance to configure it. */ static void source_setup (GstElement *pipeline, GstElement *source, CustomData *data) { GstAudioInfo info; GstCaps *audio_caps; g_print ("Source has been created. Configuring.\n"); data->app_source = source; /* Configure appsrc */ 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); 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); g_free (audio_caps_text); } int main(int argc, char *argv[]) { CustomData data; GstBus *bus; /* Initialize cumstom data structure */ memset (&data, 0, sizeof (data)); data.b = 1; /* For waveform generation */ data.d = 1; /* Initialize GStreamer */ gst_init (&argc, &argv); /* Create the playbin element */ data.pipeline = gst_parse_launch ("playbin uri=appsrc://", NULL); g_signal_connect (data.pipeline, "source-setup", G_CALLBACK (source_setup), &data); /* 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); /* Start playing the pipeline */ gst_element_set_state (data.pipeline, GST_STATE_PLAYING); /* 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); /* Free resources */ gst_element_set_state (data.pipeline, GST_STATE_NULL); gst_object_unref (data.pipeline); return 0; } ``` To use an `appsrc` as the source for the pipeline, simply instantiate a `playbin` and set its URI to `appsrc://` ``` c /* Create the playbin element */ data.pipeline = gst_parse_launch ("playbin uri=appsrc://", NULL); ``` `playbin` will create an internal `appsrc` element and fire the `source-setup` signal to allow the application to configure it: ``` c g_signal_connect (data.pipeline, "source-setup", G_CALLBACK (source_setup), &data); ``` In particular, it is important to set the caps property of `appsrc`, since, once the signal handler returns, `playbin` will instantiate the next element in the pipeline according to these caps: ``` c /* This function is called when playbin has created the appsrc element, so we have * a chance to configure it. */ static void source_setup (GstElement *pipeline, GstElement *source, CustomData *data) { GstAudioInfo info; GstCaps *audio_caps; g_print ("Source has been created. Configuring.\n"); data->app_source = source; /* Configure appsrc */ 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); 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); g_free (audio_caps_text); } ``` The configuration of the `appsrc` is exactly the same as in [](sdk-basic-tutorial-short-cutting-the-pipeline.md): the caps are set to `audio/x-raw`, and two callbacks are registered, so the element can tell the application when it needs to start and stop pushing data. See [](sdk-basic-tutorial-short-cutting-the-pipeline.md) for more details. From this point onwards, `playbin` takes care of the rest of the pipeline, and the application only needs to worry about generating more data when told so. To learn how data can be extracted from `playbin` using the `appsink` element, see [](sdk-playback-tutorial-custom-playbin-sinks.md). ## Conclusion This tutorial applies the concepts shown in [](sdk-basic-tutorial-short-cutting-the-pipeline.md) to `playbin`. In particular, it has shown: - How to connect `appsrc` with `playbin` using the special URI `appsrc://` - How to configure the `appsrc` using the `source-setup` signal It has been a pleasure having you here, and see you soon!