GStreamer Basic tutorial 8 Short-cutting the pipeline

   #include <gst/gst.h>
   #include <gst/audio/audio.h>
   #include <string.h>
   
   #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, *app_source, *tee, *audio_queue, *audio_convert1, *audio_resample, *audio_sink;
     GstElement *video_queue, *audio_convert2, *visual, *video_convert, *video_sink;
     GstElement *app_queue, *app_sink;
   
     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 idle 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 (num_samples, 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;
     }
   }
   
   /* The appsink has received a buffer */
   static GstFlowReturn new_sample (GstElement *sink, CustomData *data) {
     GstSample *sample;
   
     /* Retrieve the buffer */
     g_signal_emit_by_name (sink, "pull-sample", &sample);
     if (sample) {
       /* The only thing we do in this example is print a * to indicate a received buffer */
       g_print ("*");
       gst_sample_unref (sample);
       return GST_FLOW_OK;
     }
   
     return GST_FLOW_ERROR;
   }
   
   /* 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);
   }
   
   int main(int argc, char *argv[]) {
     CustomData data;
     GstPad *tee_audio_pad, *tee_video_pad, *tee_app_pad;
     GstPad *queue_audio_pad, *queue_video_pad, *queue_app_pad;
     GstAudioInfo info;
     GstCaps *audio_caps;
     GstBus *bus;
   
     /* Initialize custom data structure */
     memset (&data, 0, sizeof (data));
     data.b = 1; /* For waveform generation */
     data.d = 1;
   
     /* Initialize GStreamer */
     gst_init (&argc, &argv);
   
     /* Create the elements */
     data.app_source = gst_element_factory_make ("appsrc", "audio_source");
     data.tee = gst_element_factory_make ("tee", "tee");
     data.audio_queue = gst_element_factory_make ("queue", "audio_queue");
     data.audio_convert1 = gst_element_factory_make ("audioconvert", "audio_convert1");
     data.audio_resample = gst_element_factory_make ("audioresample", "audio_resample");
     data.audio_sink = gst_element_factory_make ("autoaudiosink", "audio_sink");
     data.video_queue = gst_element_factory_make ("queue", "video_queue");
     data.audio_convert2 = gst_element_factory_make ("audioconvert", "audio_convert2");
     data.visual = gst_element_factory_make ("wavescope", "visual");
     data.video_convert = gst_element_factory_make ("videoconvert", "video_convert");
     data.video_sink = gst_element_factory_make ("autovideosink", "video_sink");
     data.app_queue = gst_element_factory_make ("queue", "app_queue");
     data.app_sink = gst_element_factory_make ("appsink", "app_sink");
   
     /* Create the empty pipeline */
     data.pipeline = gst_pipeline_new ("test-pipeline");
   
     if (!data.pipeline || !data.app_source || !data.tee || !data.audio_queue || !data.audio_convert1 ||
         !data.audio_resample || !data.audio_sink || !data.video_queue || !data.audio_convert2 || !data.visual ||
         !data.video_convert || !data.video_sink || !data.app_queue || !data.app_sink) {
       g_printerr ("Not all elements could be created.\n");
       return -1;
     }
   
     /* Configure wavescope */
     g_object_set (data.visual, "shader", 0, "style", 0, NULL);
   
     /* 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 (data.app_source, "caps", audio_caps, "format", GST_FORMAT_TIME, NULL);
     g_signal_connect (data.app_source, "need-data", G_CALLBACK (start_feed), &data);
     g_signal_connect (data.app_source, "enough-data", G_CALLBACK (stop_feed), &data);
   
     /* Configure appsink */
     g_object_set (data.app_sink, "emit-signals", TRUE, "caps", audio_caps, NULL);
     g_signal_connect (data.app_sink, "new-sample", G_CALLBACK (new_sample), &data);
     gst_caps_unref (audio_caps);
   
     /* Link all elements that can be automatically linked because they have "Always" pads */
     gst_bin_add_many (GST_BIN (data.pipeline), data.app_source, data.tee, data.audio_queue, data.audio_convert1, data.audio_resample,
         data.audio_sink, data.video_queue, data.audio_convert2, data.visual, data.video_convert, data.video_sink, data.app_queue,
         data.app_sink, NULL);
     if (gst_element_link_many (data.app_source, data.tee, NULL) != TRUE ||
         gst_element_link_many (data.audio_queue, data.audio_convert1, data.audio_resample, data.audio_sink, NULL) != TRUE ||
         gst_element_link_many (data.video_queue, data.audio_convert2, data.visual, data.video_convert, data.video_sink, NULL) != TRUE ||
         gst_element_link_many (data.app_queue, data.app_sink, NULL) != TRUE) {
       g_printerr ("Elements could not be linked.\n");
       gst_object_unref (data.pipeline);
       return -1;
     }
   
     /* Manually link the Tee, which has "Request" pads */
     tee_audio_pad = gst_element_request_pad_simple (data.tee, "src_%u");
     g_print ("Obtained request pad %s for audio branch.\n", gst_pad_get_name (tee_audio_pad));
     queue_audio_pad = gst_element_get_static_pad (data.audio_queue, "sink");
     tee_video_pad = gst_element_request_pad_simple (data.tee, "src_%u");
     g_print ("Obtained request pad %s for video branch.\n", gst_pad_get_name (tee_video_pad));
     queue_video_pad = gst_element_get_static_pad (data.video_queue, "sink");
     tee_app_pad = gst_element_request_pad_simple (data.tee, "src_%u");
     g_print ("Obtained request pad %s for app branch.\n", gst_pad_get_name (tee_app_pad));
     queue_app_pad = gst_element_get_static_pad (data.app_queue, "sink");
     if (gst_pad_link (tee_audio_pad, queue_audio_pad) != GST_PAD_LINK_OK ||
         gst_pad_link (tee_video_pad, queue_video_pad) != GST_PAD_LINK_OK ||
         gst_pad_link (tee_app_pad, queue_app_pad) != GST_PAD_LINK_OK) {
       g_printerr ("Tee could not be linked\n");
       gst_object_unref (data.pipeline);
       return -1;
     }
     gst_object_unref (queue_audio_pad);
     gst_object_unref (queue_video_pad);
     gst_object_unref (queue_app_pad);
   
     /* 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);
   
     /* Release the request pads from the Tee, and unref them */
     gst_element_release_request_pad (data.tee, tee_audio_pad);
     gst_element_release_request_pad (data.tee, tee_video_pad);
     gst_element_release_request_pad (data.tee, tee_app_pad);
     gst_object_unref (tee_audio_pad);
     gst_object_unref (tee_video_pad);
     gst_object_unref (tee_app_pad);
   
     /* Free resources */
     gst_element_set_state (data.pipeline, GST_STATE_NULL);
     gst_object_unref (data.pipeline);
     return 0;
   }