This chapter will discuss how you can manipulate your pipeline in several ways from your application on. Parts of this chapter are very lowlevel, so be assured that you'll need some programming knowledge and a good understanding of &GStreamer; before you start reading this. Topics that will be discussed here include how you can insert data into a pipeline from your application, how to read data from a pipeline, how to manipulate the pipeline's speed, length, starting point and how to listen to a pipeline's data processing. Probing is best envisioned as a pad listener. Technically, a probe is nothing more than a callback that can be attached to a pad. You can attach a probe using gst_pad_add_probe (). Similarly, one can use the gst_pad_remove_probe () to remove the callback again. The probe notifies you of any activity that happens on the pad, like buffers, events and queries. You can define what kind of notifications you are interested in when you add the probe. The probe can notify you of the following activity on pads: A buffer is pushed or pulled. You want to specify the GST_PAD_PROBE_TYPE_BUFFER when registering the probe. Because the pad can be scheduled in different ways, it is possible to also specify in what scheduling mode you are interested with the optional GST_PAD_PROBE_TYPE_PUSH and GST_PAD_PROBE_TYPE_PULL flags. You can use this probe to inspect, modify or drop the buffer. See . A bufferlist is pushed. Use the GST_PAD_PROBE_TYPE_BUFFER_LIST when registering the probe. An event travels over a pad. Use the GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM and GST_PAD_PROBE_TYPE_EVENT_UPSTREAM flags to select downstream and upstream events. There is also a convenience GST_PAD_PROBE_TYPE_EVENT_BOTH to be notified of events going both upstream and downstream. By default, flush events do not cause a notification. You need to explicitly enable GST_PAD_PROBE_TYPE_EVENT_FLUSH to receive callbacks from flushing events. Events are always only notified in push mode. You can use this probe to inspect, modify or drop the event. A query travels over a pad. Use the GST_PAD_PROBE_TYPE_QUERY_DOWNSTREAM and GST_PAD_PROBE_TYPE_QUERY_UPSTREAM flags to select downstream and upstream queries. The convenience GST_PAD_PROBE_TYPE_QUERY_BOTH can also be used to select both directions. Query probes will be notified twice, once when the query travels upstream/downstream and once when the query result is returned. You can select in what stage the callback will be called with the GST_PAD_PROBE_TYPE_PUSH and GST_PAD_PROBE_TYPE_PULL, respectively when the query is performed and when the query result is returned. You can use this probe to inspect or modify the query. You can also answer the query in the probe callback by placing the result value in the query and by returning GST_PAD_PROBE_DROP from the callback. In addition to notifying you of dataflow, you can also ask the probe to block the dataflow when the callback returns. This is called a blocking probe and is activated by specifying the GST_PAD_PROBE_TYPE_BLOCK flag. You can use this flag with the other flags to only block dataflow on selected activity. A pad becomes unblocked again if you remove the probe or when you return GST_PAD_PROBE_REMOVE from the callback. You can let only the currently blocked item pass by returning GST_PAD_PROBE_PASS from the callback, it will block again on the next item. Blocking probes are used to temporarily block pads because they are unlinked or because you are going to unlink them. If the dataflow is not blocked, the pipeline would go into an error state if data is pushed on an unlinked pad. We will se how to use blocking probes to partially preroll a pipeline. See also . Be notified when no activity is happening on a pad. You install this probe with the GST_PAD_PROBE_TYPE_IDLE flag. You can specify GST_PAD_PROBE_TYPE_PUSH and/or GST_PAD_PROBE_TYPE_PULL to only be notified depending on the pad scheduling mode. The IDLE probe is also a blocking probe in that it will not let any data pass on the pad for as long as the IDLE probe is installed. You can use idle probes to dynamically relink a pad. We will see how to use idle probes to replace an element in the pipeline. See also . Data probes allow you to be notified when there is data passing on a pad. When adding the probe, specify the GST_PAD_PROBE_TYPE_BUFFER and/or GST_PAD_PROBE_TYPE_BUFFER_LIST. Data probes run in pipeline streaming thread context, so callbacks should try to not block and generally not do any weird stuff, since this could have a negative impact on pipeline performance or, in case of bugs, cause deadlocks or crashes. More precisely, one should usually not call any GUI-related functions from within a probe callback, nor try to change the state of the pipeline. An application may post custom messages on the pipeline's bus though to communicate with the main application thread and have it do things like stop the pipeline. In any case, most common buffer operations that elements can do in _chain () functions, can be done in probe callbacks as well. The example below gives a short impression on how to use them. static GstPadProbeReturn cb_have_data (GstPad *pad, GstPadProbeInfo *info, gpointer user_data) { gint x, y; GstMapInfo map; guint16 *ptr, t; GstBuffer *buffer; buffer = GST_PAD_PROBE_INFO_BUFFER (info); buffer = gst_buffer_make_writable (buffer); gst_buffer_map (buffer, &map, GST_MAP_WRITE); ptr = (guint16 *) map.data; /* invert data */ for (y = 0; y < 288; y++) { for (x = 0; x < 384 / 2; x++) { t = ptr[384 - 1 - x]; ptr[384 - 1 - x] = ptr[x]; ptr[x] = t; } ptr += 384; } gst_buffer_unmap (buffer, &map); GST_PAD_PROBE_INFO_DATA (info) = buffer; return GST_PAD_PROBE_OK; } gint main (gint argc, gchar *argv[]) { GMainLoop *loop; GstElement *pipeline, *src, *sink, *filter, *csp; GstCaps *filtercaps; GstPad *pad; /* init GStreamer */ gst_init (&argc, &argv); loop = g_main_loop_new (NULL, FALSE); /* build */ pipeline = gst_pipeline_new ("my-pipeline"); src = gst_element_factory_make ("videotestsrc", "src"); if (src == NULL) g_error ("Could not create 'videotestsrc' element"); filter = gst_element_factory_make ("capsfilter", "filter"); g_assert (filter != NULL); /* should always exist */ csp = gst_element_factory_make ("videoconvert", "csp"); if (csp == NULL) g_error ("Could not create 'videoconvert' element"); sink = gst_element_factory_make ("xvimagesink", "sink"); if (sink == NULL) { sink = gst_element_factory_make ("ximagesink", "sink"); if (sink == NULL) g_error ("Could not create neither 'xvimagesink' nor 'ximagesink' element"); } gst_bin_add_many (GST_BIN (pipeline), src, filter, csp, sink, NULL); gst_element_link_many (src, filter, csp, sink, NULL); filtercaps = gst_caps_new_simple ("video/x-raw", "format", G_TYPE_STRING, "RGB16", "width", G_TYPE_INT, 384, "height", G_TYPE_INT, 288, "framerate", GST_TYPE_FRACTION, 25, 1, NULL); g_object_set (G_OBJECT (filter), "caps", filtercaps, NULL); gst_caps_unref (filtercaps); pad = gst_element_get_static_pad (src, "src"); gst_pad_add_probe (pad, GST_PAD_PROBE_TYPE_BUFFER, (GstPadProbeCallback) cb_have_data, NULL, NULL); gst_object_unref (pad); /* run */ gst_element_set_state (pipeline, GST_STATE_PLAYING); /* wait until it's up and running or failed */ if (gst_element_get_state (pipeline, NULL, NULL, -1) == GST_STATE_CHANGE_FAILURE) { g_error ("Failed to go into PLAYING state"); } g_print ("Running ...\n"); g_main_loop_run (loop); /* exit */ gst_element_set_state (pipeline, GST_STATE_NULL); gst_object_unref (pipeline); return 0; } ]]> Compare that output with the output of gst-launch-1.0 videotestsrc ! xvimagesink, just so you know what you're looking for. Strictly speaking, a pad probe callback is only allowed to modify the buffer content if the buffer is writable. Whether this is the case or not depends a lot on the pipeline and the elements involved. Often enough, this is the case, but sometimes it is not, and if it is not then unexpected modification of the data or metadata can introduce bugs that are very hard to debug and track down. You can check if a buffer is writable with gst_buffer_is_writable (). Since you can pass back a different buffer than the one passed in, it is a good idea to make the buffer writable in the callback function with gst_buffer_make_writable (). Pad probes are suited best for looking at data as it passes through the pipeline. If you need to modify data, you should better write your own GStreamer element. Base classes like GstAudioFilter, GstVideoFilter or GstBaseTransform make this fairly easy. If you just want to inspect buffers as they pass through the pipeline, you don't even need to set up pad probes. You could also just insert an identity element into the pipeline and connect to its "handoff" signal. The identity element also provides a few useful debugging tools like the "dump" property or the "last-message" property (the latter is enabled by passing the '-v' switch to gst-launch and by setting the silent property on the identity to FALSE). In this example we will show you how to play back a region of a media file. The goal is to only play the part of a file from 2 seconds to 5 seconds and then EOS. In a first step we will set a uridecodebin element to the PAUSED state and make sure that we block all the source pads that are created. When all the source pads are blocked, we have data on all source pads and we say that the uridecodebin is prerolled. In a prerolled pipeline we can ask for the duration of the media and we can also perform seeks. We are interested in performing a seek operation on the pipeline to select the range of media that we are interested in. After we configure the region we are interested in, we can link the sink element, unblock the source pads and set the pipeline to the playing state. You will see that exactly the requested region is played by the sink before it goes to EOS. What follows What follows is an example application that loosly follows this algorithm. static GMainLoop *loop; static volatile gint counter; static GstBus *bus; static gboolean prerolled = FALSE; static GstPad *sinkpad; static void dec_counter (GstElement * pipeline) { if (prerolled) return; if (g_atomic_int_dec_and_test (&counter)) { /* all probes blocked and no-more-pads signaled, post * message on the bus. */ prerolled = TRUE; gst_bus_post (bus, gst_message_new_application ( GST_OBJECT_CAST (pipeline), gst_structure_new_empty ("ExPrerolled"))); } } /* called when a source pad of uridecodebin is blocked */ static GstPadProbeReturn cb_blocked (GstPad *pad, GstPadProbeInfo *info, gpointer user_data) { GstElement *pipeline = GST_ELEMENT (user_data); if (prerolled) return GST_PAD_PROBE_REMOVE; dec_counter (pipeline); return GST_PAD_PROBE_OK; } /* called when uridecodebin has a new pad */ static void cb_pad_added (GstElement *element, GstPad *pad, gpointer user_data) { GstElement *pipeline = GST_ELEMENT (user_data); if (prerolled) return; g_atomic_int_inc (&counter); gst_pad_add_probe (pad, GST_PAD_PROBE_TYPE_BLOCK_DOWNSTREAM, (GstPadProbeCallback) cb_blocked, pipeline, NULL); /* try to link to the video pad */ gst_pad_link (pad, sinkpad); } /* called when uridecodebin has created all pads */ static void cb_no_more_pads (GstElement *element, gpointer user_data) { GstElement *pipeline = GST_ELEMENT (user_data); if (prerolled) return; dec_counter (pipeline); } /* called when a new message is posted on the bus */ static void cb_message (GstBus *bus, GstMessage *message, gpointer user_data) { GstElement *pipeline = GST_ELEMENT (user_data); switch (GST_MESSAGE_TYPE (message)) { case GST_MESSAGE_ERROR: g_print ("we received an error!\n"); g_main_loop_quit (loop); break; case GST_MESSAGE_EOS: g_print ("we reached EOS\n"); g_main_loop_quit (loop); break; case GST_MESSAGE_APPLICATION: { if (gst_message_has_name (message, "ExPrerolled")) { /* it's our message */ g_print ("we are all prerolled, do seek\n"); gst_element_seek (pipeline, 1.0, GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE, GST_SEEK_TYPE_SET, 2 * GST_SECOND, GST_SEEK_TYPE_SET, 5 * GST_SECOND); gst_element_set_state (pipeline, GST_STATE_PLAYING); } break; } default: break; } } gint main (gint argc, gchar *argv[]) { GstElement *pipeline, *src, *csp, *vs, *sink; /* init GStreamer */ gst_init (&argc, &argv); loop = g_main_loop_new (NULL, FALSE); if (argc < 2) { g_print ("usage: %s ", argv[0]); return -1; } /* build */ pipeline = gst_pipeline_new ("my-pipeline"); bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline)); gst_bus_add_signal_watch (bus); g_signal_connect (bus, "message", (GCallback) cb_message, pipeline); src = gst_element_factory_make ("uridecodebin", "src"); if (src == NULL) g_error ("Could not create 'uridecodebin' element"); g_object_set (src, "uri", argv[1], NULL); csp = gst_element_factory_make ("videoconvert", "csp"); if (csp == NULL) g_error ("Could not create 'videoconvert' element"); vs = gst_element_factory_make ("videoscale", "vs"); if (csp == NULL) g_error ("Could not create 'videoscale' element"); sink = gst_element_factory_make ("autovideosink", "sink"); if (sink == NULL) g_error ("Could not create 'autovideosink' element"); gst_bin_add_many (GST_BIN (pipeline), src, csp, vs, sink, NULL); /* can't link src yet, it has no pads */ gst_element_link_many (csp, vs, sink, NULL); sinkpad = gst_element_get_static_pad (csp, "sink"); /* for each pad block that is installed, we will increment * the counter. for each pad block that is signaled, we * decrement the counter. When the counter is 0 we post * an app message to tell the app that all pads are * blocked. Start with 1 that is decremented when no-more-pads * is signaled to make sure that we only post the message * after no-more-pads */ g_atomic_int_set (&counter, 1); g_signal_connect (src, "pad-added", (GCallback) cb_pad_added, pipeline); g_signal_connect (src, "no-more-pads", (GCallback) cb_no_more_pads, pipeline); gst_element_set_state (pipeline, GST_STATE_PAUSED); g_main_loop_run (loop); gst_element_set_state (pipeline, GST_STATE_NULL); gst_object_unref (sinkpad); gst_object_unref (bus); gst_object_unref (pipeline); g_main_loop_unref (loop); return 0; } ]]> Note that we use a custom application message to signal the main thread that the uridecidebin is prerolled. The main thread will then issue a flushing seek to the requested region. The flush will temporarily unblock the pad and reblock them when new data arrives again. We detect this second block to remove the probes. Then we set the pipeline to PLAYING and it should play from 2 to 5 seconds, then EOS and exit the application. Many people have expressed the wish to use their own sources to inject data into a pipeline. Some people have also expressed the wish to grab the output in a pipeline and take care of the actual output inside their application. While either of these methods are strongly discouraged, &GStreamer; offers support for this. Beware! You need to know what you are doing. Since you don't have any support from a base class you need to thoroughly understand state changes and synchronization. If it doesn't work, there are a million ways to shoot yourself in the foot. It's always better to simply write a plugin and have the base class manage it. See the Plugin Writer's Guide for more information on this topic. Also see the next section, which will explain how to embed plugins statically in your application. There's two possible elements that you can use for the above-mentioned purposes. Those are called appsrc (an imaginary source) and appsink (an imaginary sink). The same method applies to each of those elements. Here, we will discuss how to use those elements to insert (using appsrc) or grab (using appsink) data from a pipeline, and how to set negotiation. Both appsrc and appsink provide 2 sets of API. One API uses standard GObject (action) signals and properties. The same API is also available as a regular C api. The C api is more performant but requires you to link to the app library in order to use the elements. First we look at some examples for appsrc, which lets you insert data into the pipeline from the application. Appsrc has some configuration options that define how it will operate. You should decide about the following configurations: Will the appsrc operate in push or pull mode. The stream-type property can be used to control this. stream-type of random-access will activate pull mode scheduling while the other stream-types activate push mode. The caps of the buffers that appsrc will push out. This needs to be configured with the caps property. The caps must be set to a fixed caps and will be used to negotiate a format downstream. It the appsrc operates in live mode or not. This can be configured with the is-live property. When operating in live-mode it is important to configure the min-latency and max-latency in appsrc. The min-latency should be set to the amount of time it takes between capturing a buffer and when it is pushed inside appsrc. In live mode, you should timestamp the buffers with the pipeline running-time when the first byte of the buffer was captured before feeding them to appsrc. You can let appsrc do the timestaping with the do-timestamp property (but then the min-latency must be set to 0 because it timestamps based on the running-time when the buffer entered appsrc). The format of the SEGMENT event that appsrc will push. The format has implications for how the running-time of the buffers will be calculated so you must be sure you understand this. For live sources you probably want to set the format property to GST_FORMAT_TIME. For non-live source it depends on the media type that you are handling. If you plan to timestamp the buffers, you should probably put a GST_FORMAT_TIME format, otherwise GST_FORMAT_BYTES might be appropriate. If appsrc operates in random-access mode, it is important to configure the size property of appsrc with the number of bytes in the stream. This will allow downstream elements to know the size of the media and alows them to seek to the end of the stream when needed. The main way of handling data to appsrc is by using the function gst_app_src_push_buffer () or by emiting the push-buffer action signal. This will put the buffer onto a queue from which appsrc will read from in its streaming thread. It is important to note that data transport will not happen from the thread that performed the push-buffer call. The max-bytes property controls how much data can be queued in appsrc before appsrc considers the queue full. A filled internal queue will always signal the enough-data signal, which signals the application that it should stop pushing data into appsrc. The block property will cause appsrc to block the push-buffer method until free data becomes available again. When the internal queue is running out of data, the need-data signal is emitted, which signals the application that it should start pushing more data into appsrc. In addition to the need-data and enough-data signals, appsrc can emit the seek-data signal when the stream-mode property is set to seekable or random-access. The signal argument will contain the new desired position in the stream expressed in the unit set with the format property. After receiving the seek-data signal, the application should push-buffers from the new position. When the last byte is pushed into appsrc, you must call gst_app_src_end_of_stream () to make it send an EOS downstream. These signals allow the application to operate appsrc in push and pull mode as will be explained next. When appsrc is configured in push mode (stream-type is stream or seekable), the application repeatedly calls the push-buffer method with a new buffer. Optionally, the queue size in the appsrc can be controlled with the enough-data and need-data signals by respectively stopping/starting the push-buffer calls. The value of the min-percent property defines how empty the internal appsrc queue needs to be before the need-data signal will be fired. You can set this to some value >0 to avoid completely draining the queue. When the stream-type is set to seekable, don't forget to implement a seek-data callback. Use this model when implementing various network protocols or hardware devices. In the pull model, data is fed to appsrc from the need-data signal handler. You should push exactly the amount of bytes requested in the need-data signal. You are only allowed to push less bytes when you are at the end of the stream. Use this model for file access or other randomly accessable sources. This example application will generate black/white (it switches every second) video to an Xv-window output by using appsrc as a source with caps to force a format. We use a colorspace conversion element to make sure that we feed the right format to your X server. We configure a video stream with a variable framerate (0/1) and we set the timestamps on the outgoing buffers in such a way that we play 2 frames per second. Note how we use the pull mode method of pushing new buffers into appsrc although appsrc is running in push mode. static GMainLoop *loop; static void cb_need_data (GstElement *appsrc, guint unused_size, gpointer user_data) { static gboolean white = FALSE; static GstClockTime timestamp = 0; GstBuffer *buffer; guint size; GstFlowReturn ret; size = 385 * 288 * 2; buffer = gst_buffer_new_allocate (NULL, size, NULL); /* this makes the image black/white */ gst_buffer_memset (buffer, 0, white ? 0xff : 0x0, size); white = !white; GST_BUFFER_PTS (buffer) = timestamp; GST_BUFFER_DURATION (buffer) = gst_util_uint64_scale_int (1, GST_SECOND, 2); timestamp += GST_BUFFER_DURATION (buffer); g_signal_emit_by_name (appsrc, "push-buffer", buffer, &ret); if (ret != GST_FLOW_OK) { /* something wrong, stop pushing */ g_main_loop_quit (loop); } } gint main (gint argc, gchar *argv[]) { GstElement *pipeline, *appsrc, *conv, *videosink; /* init GStreamer */ gst_init (&argc, &argv); loop = g_main_loop_new (NULL, FALSE); /* setup pipeline */ pipeline = gst_pipeline_new ("pipeline"); appsrc = gst_element_factory_make ("appsrc", "source"); conv = gst_element_factory_make ("videoconvert", "conv"); videosink = gst_element_factory_make ("xvimagesink", "videosink"); /* setup */ g_object_set (G_OBJECT (appsrc), "caps", gst_caps_new_simple ("video/x-raw", "format", G_TYPE_STRING, "RGB16", "width", G_TYPE_INT, 384, "height", G_TYPE_INT, 288, "framerate", GST_TYPE_FRACTION, 0, 1, NULL), NULL); gst_bin_add_many (GST_BIN (pipeline), appsrc, conv, videosink, NULL); gst_element_link_many (appsrc, conv, videosink, NULL); /* setup appsrc */ g_object_set (G_OBJECT (appsrc), "stream-type", 0, "format", GST_FORMAT_TIME, NULL); g_signal_connect (appsrc, "need-data", G_CALLBACK (cb_need_data), NULL); /* play */ gst_element_set_state (pipeline, GST_STATE_PLAYING); g_main_loop_run (loop); /* clean up */ gst_element_set_state (pipeline, GST_STATE_NULL); gst_object_unref (GST_OBJECT (pipeline)); g_main_loop_unref (loop); return 0; } ]]> Unlike appsrc, appsink is a little easier to use. It also supports a pull and push based model of getting data from the pipeline. The normal way of retrieving samples from appsink is by using the gst_app_sink_pull_sample() and gst_app_sink_pull_preroll() methods or by using the pull-sample and pull-preroll signals. These methods block until a sample becomes available in the sink or when the sink is shut down or reaches EOS. Appsink will internally use a queue to collect buffers from the streaming thread. If the application is not pulling samples fast enough, this queue will consume a lot of memory over time. The max-buffers property can be used to limit the queue size. The drop property controls whether the streaming thread blocks or if older buffers are dropped when the maximum queue size is reached. Note that blocking the streaming thread can negatively affect real-time performance and should be avoided. If a blocking behaviour is not desirable, setting the emit-signals property to TRUE will make appsink emit the new-sample and new-preroll signals when a sample can be pulled without blocking. The caps property on appsink can be used to control the formats that appsink can receive. This property can contain non-fixed caps, the format of the pulled samples can be obtained by getting the sample caps. If one of the pull-preroll or pull-sample methods return NULL, the appsink is stopped or in the EOS state. You can check for the EOS state with the eos property or with the gst_app_sink_is_eos() method. The eos signal can also be used to be informed when the EOS state is reached to avoid polling. Consider configuring the following properties in the appsink: The sync property if you want to have the sink base class synchronize the buffer against the pipeline clock before handing you the sample. Enable Quality-of-Service with the qos property. If you are dealing with raw video frames and let the base class sycnhronize on the clock, it might be a good idea to also let the base class send QOS events upstream. The caps property that contains the accepted caps. Upstream elements will try to convert the format so that it matches the configured caps on appsink. You must still check the GstSample to get the actual caps of the buffer. What follows is an example on how to capture a snapshot of a video stream using appsink. #ifdef HAVE_GTK #include #endif #include #define CAPS "video/x-raw,format=RGB,width=160,pixel-aspect-ratio=1/1" int main (int argc, char *argv[]) { GstElement *pipeline, *sink; gint width, height; GstSample *sample; gchar *descr; GError *error = NULL; gint64 duration, position; GstStateChangeReturn ret; gboolean res; GstMapInfo map; gst_init (&argc, &argv); if (argc != 2) { g_print ("usage: %s \n Writes snapshot.png in the current directory\n", argv[0]); exit (-1); } /* create a new pipeline */ descr = g_strdup_printf ("uridecodebin uri=%s ! videoconvert ! videoscale ! " " appsink name=sink caps=\"" CAPS "\"", argv[1]); pipeline = gst_parse_launch (descr, &error); if (error != NULL) { g_print ("could not construct pipeline: %s\n", error->message); g_error_free (error); exit (-1); } /* get sink */ sink = gst_bin_get_by_name (GST_BIN (pipeline), "sink"); /* set to PAUSED to make the first frame arrive in the sink */ ret = gst_element_set_state (pipeline, GST_STATE_PAUSED); switch (ret) { case GST_STATE_CHANGE_FAILURE: g_print ("failed to play the file\n"); exit (-1); case GST_STATE_CHANGE_NO_PREROLL: /* for live sources, we need to set the pipeline to PLAYING before we can * receive a buffer. We don't do that yet */ g_print ("live sources not supported yet\n"); exit (-1); default: break; } /* This can block for up to 5 seconds. If your machine is really overloaded, * it might time out before the pipeline prerolled and we generate an error. A * better way is to run a mainloop and catch errors there. */ ret = gst_element_get_state (pipeline, NULL, NULL, 5 * GST_SECOND); if (ret == GST_STATE_CHANGE_FAILURE) { g_print ("failed to play the file\n"); exit (-1); } /* get the duration */ gst_element_query_duration (pipeline, GST_FORMAT_TIME, &duration); if (duration != -1) /* we have a duration, seek to 5% */ position = duration * 5 / 100; else /* no duration, seek to 1 second, this could EOS */ position = 1 * GST_SECOND; /* seek to the a position in the file. Most files have a black first frame so * by seeking to somewhere else we have a bigger chance of getting something * more interesting. An optimisation would be to detect black images and then * seek a little more */ gst_element_seek_simple (pipeline, GST_FORMAT_TIME, GST_SEEK_FLAG_KEY_UNIT | GST_SEEK_FLAG_FLUSH, position); /* get the preroll buffer from appsink, this block untils appsink really * prerolls */ g_signal_emit_by_name (sink, "pull-preroll", &sample, NULL); /* if we have a buffer now, convert it to a pixbuf. It's possible that we * don't have a buffer because we went EOS right away or had an error. */ if (sample) { GstBuffer *buffer; GstCaps *caps; GstStructure *s; /* get the snapshot buffer format now. We set the caps on the appsink so * that it can only be an rgb buffer. The only thing we have not specified * on the caps is the height, which is dependant on the pixel-aspect-ratio * of the source material */ caps = gst_sample_get_caps (sample); if (!caps) { g_print ("could not get snapshot format\n"); exit (-1); } s = gst_caps_get_structure (caps, 0); /* we need to get the final caps on the buffer to get the size */ res = gst_structure_get_int (s, "width", &width); res |= gst_structure_get_int (s, "height", &height); if (!res) { g_print ("could not get snapshot dimension\n"); exit (-1); } /* create pixmap from buffer and save, gstreamer video buffers have a stride * that is rounded up to the nearest multiple of 4 */ buffer = gst_sample_get_buffer (sample); gst_buffer_map (buffer, &map, GST_MAP_READ); #ifdef HAVE_GTK pixbuf = gdk_pixbuf_new_from_data (map.data, GDK_COLORSPACE_RGB, FALSE, 8, width, height, GST_ROUND_UP_4 (width * 3), NULL, NULL); /* save the pixbuf */ gdk_pixbuf_save (pixbuf, "snapshot.png", "png", &error, NULL); #endif gst_buffer_unmap (buffer, &map); } else { g_print ("could not make snapshot\n"); } /* cleanup and exit */ gst_element_set_state (pipeline, GST_STATE_NULL); gst_object_unref (pipeline); exit (0); } ]]> Sometimes you'll want to set a specific format, for example a video size and format or an audio bitsize and number of channels. You can do this by forcing a specific GstCaps on the pipeline, which is possible by using filtered caps. You can set a filtered caps on a link by using the capsfilter element in between the two elements, and specifying a GstCaps as caps property on this element. It will then only allow types matching that specified capability set for negotiation. See also . It is also possible to dynamically change the format in a pipeline while PLAYING. This can simply be done by changing the caps property on a capsfilter. The capsfilter will send a RECONFIGURE event upstream that will make the upstream element attempt to renegotiate a new format and allocator. This only works if the upstream element is not using fixed caps on the source pad. Below is an example of how you can change the caps of a pipeline while in the PLAYING state: #include #define MAX_ROUND 100 int main (int argc, char **argv) { GstElement *pipe, *filter; GstCaps *caps; gint width, height; gint xdir, ydir; gint round; GstMessage *message; gst_init (&argc, &argv); pipe = gst_parse_launch_full ("videotestsrc ! capsfilter name=filter ! " "ximagesink", NULL, GST_PARSE_FLAG_NONE, NULL); g_assert (pipe != NULL); filter = gst_bin_get_by_name (GST_BIN (pipe), "filter"); g_assert (filter); width = 320; height = 240; xdir = ydir = -10; for (round = 0; round < MAX_ROUND; round++) { gchar *capsstr; g_print ("resize to %dx%d (%d/%d) \r", width, height, round, MAX_ROUND); /* we prefer our fixed width and height but allow other dimensions to pass * as well */ capsstr = g_strdup_printf ("video/x-raw, width=(int)%d, height=(int)%d", width, height); caps = gst_caps_from_string (capsstr); g_free (capsstr); g_object_set (filter, "caps", caps, NULL); gst_caps_unref (caps); if (round == 0) gst_element_set_state (pipe, GST_STATE_PLAYING); width += xdir; if (width >= 320) xdir = -10; else if (width < 200) xdir = 10; height += ydir; if (height >= 240) ydir = -10; else if (height < 150) ydir = 10; message = gst_bus_poll (GST_ELEMENT_BUS (pipe), GST_MESSAGE_ERROR, 50 * GST_MSECOND); if (message) { g_print ("got error \n"); gst_message_unref (message); } } g_print ("done \n"); gst_object_unref (filter); gst_element_set_state (pipe, GST_STATE_NULL); gst_object_unref (pipe); return 0; } ]]> Note how we use gst_bus_poll() with a small timeout to get messages and also introduce a short sleep. It is possible to set multiple caps for the capsfilter separated with a ;. The capsfilter will try to renegotiate to the first possible format from the list. In this section we talk about some techniques for dynamically modifying the pipeline. We are talking specifically about changing the pipeline while it is in the PLAYING state without interrupting the flow. There are some important things to consider when building dynamic pipelines: When removing elements from the pipeline, make sure that there is no dataflow on unlinked pads because that will cause a fatal pipeline error. Always block source pads (in push mode) or sink pads (in pull mode) before unlinking pads. See also . When adding elements to a pipeline, make sure to put the element into the right state, usually the same state as the parent, before allowing dataflow the element. When an element is newly created, it is in the NULL state and will return an error when it receives data. See also . When adding elements to a pipeline, &GStreamer; will by default set the clock and base-time on the element to the current values of the pipeline. This means that the element will be able to construct the same pipeline running-time as the other elements in the pipeline. This means that sinks will synchronize buffers like the other sinks in the pipeline and that sources produce buffers with a running-time that matches the other sources. When unlinking elements from an upstream chain, always make sure to flush any queued data in the element by sending an EOS event down the element sink pad(s) and by waiting that the EOS leaves the elements (with an event probe). See also . A live source will produce buffers with a running-time of the current running-time in the pipeline. A pipeline without a live source produces buffers with a running-time starting from 0. Likewise, after a flushing seek, those pipelines reset the running-time back to 0. The running-time can be changed with gst_pad_set_offset (). It is important to know the running-time of the elements in the pipeline in order to maintain synchronization. Adding elements might change the state of the pipeline. Adding a non-prerolled sink, for example, brings the pipeline back to the prerolling state. Removing a non-prerolled sink, for example, might change the pipeline to PAUSED and PLAYING state. Adding a live source cancels the preroll stage and put the pipeline to the playing state. Adding a live source or other live elements might also change the latency of a pipeline. Adding or removing elements to the pipeline might change the clock selection of the pipeline. If the newly added element provides a clock, it might be worth changing the clock in the pipeline to the new clock. If, on the other hand, the element that provides the clock for the pipeline is removed, a new clock has to be selected. Adding and removing elements might cause upstream or downstream elements to renegotiate caps and or allocators. You don't really need to do anything from the application, plugins largely adapt themself to the new pipeline topology in order to optimize their formats and allocation strategy. What is important is that when you add, remove or change elements in the pipeline, it is possible that the pipeline needs to negotiate a new format and this can fail. Usually you can fix this by inserting the right converter elements where needed. See also . &GStreamer; offers support for doing about any dynamic pipeline modification but it requires you to know a bit of details before you can do this without causing pipeline errors. In the following sections we will demonstrate a couple of typical use-cases. In the next example we look at the following chain of elements: - ----. .----------. .---- - element1 | | element2 | | element3 src -> sink src -> sink - ----' '----------' '---- - We want to change element2 by element4 while the pipeline is in the PLAYING state. Let's say that element2 is a visualization and that you want to switch the visualization in the pipeline. We can't just unlink element2's sinkpad from element1's source pad because that would leave element1's source pad unlinked and would cause a streaming error in the pipeline when data is pushed on the source pad. The technique is to block the dataflow from element1's source pad before we change element2 by element4 and then resume dataflow as shown in the following steps: Block element1's source pad with a blocking pad probe. When the pad is blocked, the probe callback will be called. Inside the block callback nothing is flowing between element1 and element2 and nothing will flow until unblocked. Unlink element1 and element2. Make sure data is flushed out of element2. Some elements might internally keep some data, you need to make sure not to lose data by forcing it out of element2. You can do this by pushing EOS into element2, like this: Put an event probe on element2's source pad. Send EOS to element2's sinkpad. This makes sure the all the data inside element2 is forced out. Wait for the EOS event to appear on element2's source pad. When the EOS is received, drop it and remove the event probe. Unlink element2 and element3. You can now also remove element2 from the pipeline and set the state to NULL. Add element4 to the pipeline, if not already added. Link element4 and element3. Link element1 and element4. Make sure element4 is in the same state as the rest of the elements in the pipeline. It should be at least in the PAUSED state before it can receive buffers and events. Unblock element1's source pad probe. This will let new data into element4 and continue streaming. The above algorithm works when the source pad is blocked, i.e. when there is dataflow in the pipeline. If there is no dataflow, there is also no point in changing the element (just yet) so this algorithm can be used in the PAUSED state as well. Let show you how this works with an example. This example changes the video effect on a simple pipeline every second. static gchar *opt_effects = NULL; #define DEFAULT_EFFECTS "identity,exclusion,navigationtest," \ "agingtv,videoflip,vertigotv,gaussianblur,shagadelictv,edgetv" static GstPad *blockpad; static GstElement *conv_before; static GstElement *conv_after; static GstElement *cur_effect; static GstElement *pipeline; static GQueue effects = G_QUEUE_INIT; static GstPadProbeReturn event_probe_cb (GstPad * pad, GstPadProbeInfo * info, gpointer user_data) { GMainLoop *loop = user_data; GstElement *next; if (GST_EVENT_TYPE (GST_PAD_PROBE_INFO_DATA (info)) != GST_EVENT_EOS) return GST_PAD_PROBE_OK; gst_pad_remove_probe (pad, GST_PAD_PROBE_INFO_ID (info)); /* push current event back into the queue */ g_queue_push_tail (&effects, gst_object_ref (cur_effect)); /* take next effect from the queue */ next = g_queue_pop_head (&effects); if (next == NULL) { GST_DEBUG_OBJECT (pad, "no more effects"); g_main_loop_quit (loop); return GST_PAD_PROBE_DROP; } g_print ("Switching from '%s' to '%s'..\n", GST_OBJECT_NAME (cur_effect), GST_OBJECT_NAME (next)); gst_element_set_state (cur_effect, GST_STATE_NULL); /* remove unlinks automatically */ GST_DEBUG_OBJECT (pipeline, "removing %" GST_PTR_FORMAT, cur_effect); gst_bin_remove (GST_BIN (pipeline), cur_effect); GST_DEBUG_OBJECT (pipeline, "adding %" GST_PTR_FORMAT, next); gst_bin_add (GST_BIN (pipeline), next); GST_DEBUG_OBJECT (pipeline, "linking.."); gst_element_link_many (conv_before, next, conv_after, NULL); gst_element_set_state (next, GST_STATE_PLAYING); cur_effect = next; GST_DEBUG_OBJECT (pipeline, "done"); return GST_PAD_PROBE_DROP; } static GstPadProbeReturn pad_probe_cb (GstPad * pad, GstPadProbeInfo * info, gpointer user_data) { GstPad *srcpad, *sinkpad; GST_DEBUG_OBJECT (pad, "pad is blocked now"); /* remove the probe first */ gst_pad_remove_probe (pad, GST_PAD_PROBE_INFO_ID (info)); /* install new probe for EOS */ srcpad = gst_element_get_static_pad (cur_effect, "src"); gst_pad_add_probe (srcpad, GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM, event_probe_cb, user_data, NULL); gst_object_unref (srcpad); /* push EOS into the element, the probe will be fired when the * EOS leaves the effect and it has thus drained all of its data */ sinkpad = gst_element_get_static_pad (cur_effect, "sink"); gst_pad_send_event (sinkpad, gst_event_new_eos ()); gst_object_unref (sinkpad); return GST_PAD_PROBE_OK; } static gboolean timeout_cb (gpointer user_data) { gst_pad_add_probe (blockpad, GST_PAD_PROBE_TYPE_BLOCK_DOWNSTREAM, pad_probe_cb, user_data, NULL); return TRUE; } static gboolean bus_cb (GstBus * bus, GstMessage * msg, gpointer user_data) { GMainLoop *loop = user_data; switch (GST_MESSAGE_TYPE (msg)) { case GST_MESSAGE_ERROR:{ GError *err = NULL; gchar *dbg; gst_message_parse_error (msg, &err, &dbg); gst_object_default_error (msg->src, err, dbg); g_error_free (err); g_free (dbg); g_main_loop_quit (loop); break; } default: break; } return TRUE; } int main (int argc, char **argv) { GOptionEntry options[] = { {"effects", 'e', 0, G_OPTION_ARG_STRING, &opt_effects, "Effects to use (comma-separated list of element names)", NULL}, {NULL} }; GOptionContext *ctx; GError *err = NULL; GMainLoop *loop; GstElement *src, *sink; gchar **effect_names, **e; ctx = g_option_context_new (""); g_option_context_add_main_entries (ctx, options, NULL); g_option_context_add_group (ctx, gst_init_get_option_group ()); if (!g_option_context_parse (ctx, &argc, &argv, &err)) { g_print ("Error initializing: %s\n", err->message); return 1; } g_option_context_free (ctx); if (opt_effects != NULL) effect_names = g_strsplit (opt_effects, ",", -1); else effect_names = g_strsplit (DEFAULT_EFFECTS, ",", -1); for (e = effect_names; e != NULL && *e != NULL; ++e) { GstElement *el; el = gst_element_factory_make (*e, NULL); if (el) { g_print ("Adding effect '%s'\n", *e); g_queue_push_tail (&effects, el); } } pipeline = gst_pipeline_new ("pipeline"); src = gst_element_factory_make ("videotestsrc", NULL); g_object_set (src, "is-live", TRUE, NULL); blockpad = gst_element_get_static_pad (src, "src"); conv_before = gst_element_factory_make ("videoconvert", NULL); cur_effect = g_queue_pop_head (&effects); conv_after = gst_element_factory_make ("videoconvert", NULL); sink = gst_element_factory_make ("ximagesink", NULL); gst_bin_add_many (GST_BIN (pipeline), src, conv_before, cur_effect, conv_after, sink, NULL); gst_element_link_many (src, conv_before, cur_effect, conv_after, sink, NULL); gst_element_set_state (pipeline, GST_STATE_PLAYING); loop = g_main_loop_new (NULL, FALSE); gst_bus_add_watch (GST_ELEMENT_BUS (pipeline), bus_cb, loop); g_timeout_add_seconds (1, timeout_cb, loop); g_main_loop_run (loop); gst_element_set_state (pipeline, GST_STATE_NULL); gst_object_unref (pipeline); return 0; } ]]> Note how we added videoconvert elements before and after the effect. This is needed because some elements might operate in different colorspaces than other elements. By inserting the conversion elements you ensure that the right format can be negotiated at any time.