Snapshot with Gstreamer without EOS - c

I'm trying to take several snapshots from a source using gstreamer. With the following code, I succeed to take 9 files but with an EOS from the source (that is actually normal, it's cause by the num-buffers argument):
#include <gst/gst.h>
/* Structure to contain all our information, so we can pass it to callbacks */
typedef struct _CustomData {
GstElement *pipeline;
GstElement *source;
GstElement *convert;
GstElement *sink;
GstElement *encode;
} CustomData;
int main(int argc, char *argv[]) {
CustomData data;
GstBus *bus;
GstMessage *msg;
GstStateChangeReturn ret;
gboolean terminate = FALSE;
/* Initialize GStreamer */
gst_init (&argc, &argv);
/* Create the elements */
data.source = gst_element_factory_make ("videotestsrc", "source");
data.convert = gst_element_factory_make ("ffmpegcolorspace", "convert");
data.encode = gst_element_factory_make ("ffenc_pgm", "encode");
data.sink = gst_element_factory_make ("multifilesink", "sink");
/* Create the empty pipeline */
data.pipeline = gst_pipeline_new ("test-pipeline");
if (!data.pipeline || !data.source || !data.convert || !data.sink) {
g_printerr ("Not all elements could be created.\n");
return -1;
}
/* Build the pipeline. Note that we are NOT linking the source at this
* point. We will do it later. */
gst_bin_add_many (GST_BIN (data.pipeline), data.source, data.convert , data.encode, data.sink, NULL);
if (!gst_element_link_many (data.source, data.convert, data.encode, data.sink, NULL)) {
g_printerr ("Elements could not be linked.\n");
gst_object_unref (data.pipeline);
return -1;
}
/* Modify the source's properties */
g_object_set (data.source, "pattern", 0, NULL);
g_object_set (data.source, "num-buffers", 9, NULL);
g_object_set(data.sink, "location", "frame%05d.pgm", NULL);
/* Start playing */
ret = gst_element_set_state (data.pipeline, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE) {
g_printerr ("Unable to set the pipeline to the playing state.\n");
gst_object_unref (data.pipeline);
return -1;
}
/* Wait until error or EOS */
bus = gst_element_get_bus (data.pipeline);
msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE, GST_MESSAGE_ERROR | GST_MESSAGE_EOS);
/* Parse message */
if (msg != NULL) {
GError *err;
gchar *debug_info;
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_ERROR:
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);
break;
case GST_MESSAGE_EOS:
g_print ("End-Of-Stream reached.\n");
break;
default:
/* We should not reach here because we only asked for ERRORs and EOS */
g_printerr ("Unexpected message received.\n");
break;
}
gst_message_unref (msg);
}
/* Free resources */
gst_object_unref (bus);
gst_element_set_state (data.pipeline, GST_STATE_NULL);
gst_object_unref (data.pipeline);
return 0;
}
But my problem is that I want to continue the live after those 9 snapshots. I look for in the tee and queue capabilities but I'm not able to do anything. I think I have to do a dynamical pipeline with a multifilesink element that I paused and played but how to tell it to do only 9 files ? (max-files=9 doesn't work cause the files generated are overwritten)
Thanks

Sure, you need to add probe to count buffers and remove some elements once you don't need them.
I added few fields to your struct:
int count;
GstPad *blockpad;
GstElement *fakesink;
I created one more sink to replace end of pipeline once we saved 9 snapshots:
data.fakesink = gst_element_factory_make ("fakesink", "fakesink");
I added probe to srcpad of data.convert:
data.count = 0;
data.blockpad = gst_element_get_static_pad (data.convert, "src");
gst_pad_add_probe (data.blockpad, GST_PAD_PROBE_TYPE_BLOCK_DOWNSTREAM | GST_PAD_PROBE_TYPE_BUFFER,
pad_probe_cb, &data, NULL);
I used GStreamer 1.x so I replaced ffenc_pgm element with avenc_pgm and ffmpegcolorspace element with identity:
#include <stdio.h>
#include <gst/gst.h>
/* Structure to contain all our information, so we can pass it to callbacks */
typedef struct _CustomData {
int count;
GstPad *blockpad;
GstElement *pipeline;
GstElement *source;
GstElement *convert;
GstElement *sink;
GstElement *fakesink;
GstElement *encode;
} CustomData;
static GstPadProbeReturn
pad_probe_cb (GstPad * pad, GstPadProbeInfo * info, gpointer user_data) {
CustomData *data = user_data;
data->count++;
printf("%d\n", data->count);
if (data->count > 9)
{
gst_element_set_state (data->encode, GST_STATE_NULL);
gst_bin_remove (GST_BIN (data->pipeline), data->encode);
gst_element_set_state (data->sink, GST_STATE_NULL);
gst_bin_remove (GST_BIN (data->pipeline), data->sink);
gst_bin_add (GST_BIN (data->pipeline), data->fakesink);
gst_element_link (data->convert, data->fakesink);
gst_element_set_state (data->fakesink, GST_STATE_PLAYING);
gst_pad_remove_probe (pad, GST_PAD_PROBE_INFO_ID (info));
return GST_PAD_PROBE_REMOVE;
}
else
return GST_PAD_PROBE_PASS;
}
int main(int argc, char *argv[]) {
CustomData data;
GstBus *bus;
GstMessage *msg;
GstStateChangeReturn ret;
gboolean terminate = FALSE;
/* Initialize GStreamer */
gst_init (&argc, &argv);
/* Create the elements */
data.source = gst_element_factory_make ("videotestsrc", "source");
data.convert = gst_element_factory_make ("identity", "convert");
data.encode = gst_element_factory_make ("avenc_pgm", "encode");
data.sink = gst_element_factory_make ("multifilesink", "sink");
data.fakesink = gst_element_factory_make ("fakesink", "fakesink");
/* Create the empty pipeline */
data.pipeline = gst_pipeline_new ("test-pipeline");
if (!data.pipeline || !data.source || !data.convert || !data.sink) {
g_printerr ("Not all elements could be created.\n");
return -1;
}
/* Build the pipeline. Note that we are NOT linking the source at this
* point. We will do it later. */
gst_bin_add_many (GST_BIN (data.pipeline), data.source, data.convert , data.encode, data.sink, NULL);
if (!gst_element_link_many (data.source, data.convert, data.encode, data.sink, NULL)) {
g_printerr ("Elements could not be linked.\n");
gst_object_unref (data.pipeline);
return -1;
}
/* Modify the source's properties */
g_object_set (data.source, "pattern", 0, NULL);
g_object_set (data.source, "num-buffers", 20, NULL);
g_object_set (data.sink, "location", "frame%05d.pgm", NULL);
data.count = 0;
data.blockpad = gst_element_get_static_pad (data.convert, "src");
gst_pad_add_probe (data.blockpad, GST_PAD_PROBE_TYPE_BLOCK_DOWNSTREAM | GST_PAD_PROBE_TYPE_BUFFER,
pad_probe_cb, &data, NULL);
/* Start playing */
ret = gst_element_set_state (data.pipeline, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE) {
g_printerr ("Unable to set the pipeline to the playing state.\n");
gst_object_unref (data.pipeline);
return -1;
}
/* Wait until error or EOS */
bus = gst_element_get_bus (data.pipeline);
msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE, GST_MESSAGE_ERROR | GST_MESSAGE_EOS);
/* Parse message */
if (msg != NULL) {
GError *err;
gchar *debug_info;
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_ERROR:
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);
break;
case GST_MESSAGE_EOS:
g_print ("End-Of-Stream reached.\n");
break;
default:
/* We should not reach here because we only asked for ERRORs and EOS */
g_printerr ("Unexpected message received.\n");
break;
}
gst_message_unref (msg);
}
/* Free resources */
gst_object_unref (bus);
gst_element_set_state (data.pipeline, GST_STATE_NULL);
gst_object_unref (data.pipeline);
return 0;
}

Related

problem to play media with gstreamer and srt protocol, language c

I'm new to gstreamer and I'm trying to output the video on another port (with the srt protocol). So far I have done this and it doesn't work:
typedef struct _CustomData
{
GstElement *pipeline;
GstElement *source;
GstElement *sink;
} CustomData;
static void pad_added_handler (GstElement * src, GstPad * pad, CustomData * data);
int main (int argc, char *argv[]) {
CustomData data;
GstBus *bus;
GstMessage *msg;
GstStateChangeReturn ret;
gboolean terminate = FALSE;
/* Initialize GStreamer */
gst_init (&argc, &argv);
/* Create the elements */
data.source = gst_element_factory_make ("uridecodebin", "source");
data.sink = gst_element_make_from_uri (GST_URI_SINK,"srt://my_uri", NULL, NULL);
/* Create the empty pipeline */
data.pipeline = gst_pipeline_new ("test-pipeline");
if (!data.pipeline || !data.source || !data.sink) {
g_printerr ("Not all elements could be created.\n");
return -1;
}
/* Build the pipeline. Note that we are NOT linking the source at this point. We will do it later. */
gst_bin_add_many (GST_BIN (data.pipeline), data.source, data.sink, NULL);
/* Set the URI to play */
g_object_set (data.source, "uri", "srt://my_uri", NULL);
/* Connect to the pad-added signal */
g_signal_connect (data.source, "pad-added", G_CALLBACK (pad_added_handler), &data);
/* Start playing */
ret = gst_element_set_state (data.pipeline, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE) {
g_printerr ("Unable to set the pipeline to the playing state.\n");
gst_object_unref (data.pipeline);
return -1;
}
/* Listen to the bus */
bus = gst_element_get_bus (data.pipeline);
do {
msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE,
GST_MESSAGE_STATE_CHANGED | GST_MESSAGE_ERROR | GST_MESSAGE_EOS);
...
} while (!terminate);
/* Free resources */
gst_object_unref (bus);
gst_element_set_state (data.pipeline, GST_STATE_NULL);
gst_object_unref (data.pipeline);
return 0;
}
and pad_added_handler, the function will be called by the padd-added signal:
/* This function will be called by the pad-added signal */
static void pad_added_handler (GstElement * src, GstPad * new_pad, CustomData * data)
{
GstPad *sink_pad = gst_element_get_static_pad (data->sink, "sink");
GstPadLinkReturn ret;
GstCaps *new_pad_caps = NULL;
GstStructure *new_pad_struct = NULL;
const gchar *new_pad_type = NULL;
g_print ("Received new pad '%s' from '%s':\n", GST_PAD_NAME (new_pad), GST_ELEMENT_NAME (src));
/* If our converter is already linked, we have nothing to do here */
if (gst_pad_is_linked (sink_pad)) {
g_print ("We are already linked. Ignoring.\n");
goto exit;
}
/* Check the new pad's type */
new_pad_caps = gst_pad_get_current_caps (new_pad);
new_pad_struct = gst_caps_get_structure (new_pad_caps, 0);
new_pad_type = gst_structure_get_name (new_pad_struct);
if (!g_str_has_prefix (new_pad_type, "video/x-raw")) {
g_print ("It has type '%s' which is not raw audio. Ignoring.\n",new_pad_type);
goto exit;
}
/* Attempt the link */
ret = gst_pad_link (new_pad, sink_pad);
if (GST_PAD_LINK_FAILED (ret)) {
g_print ("Type is '%s' but link failed.\n", new_pad_type);
} else {
g_print ("Link succeeded (type '%s').\n", new_pad_type);
}
exit:
/* Unreference the new pad's caps, if we got them */
if (new_pad_caps != NULL)
gst_caps_unref (new_pad_caps);
/* Unreference the sink pad */
gst_object_unref (sink_pad);
}
I don't get any error when running the code but when I try to read the media I get the following error: "Operation not supported: Invalid socket ID"
Thank you in advance for your help

Need an example to retrieve stream statistics of output stream in Gstreamer C code?

I'm new to C coding and I am writing a basic transcoding program for a project I am working on. I was wondering if anyone has a basic example which would allow me to capture the output statistics for example (actual bitrate for video and audio, framerate, resolution size, video h264 level, etc)
Please see below code:
#include <gst/gst.h>
#include <glib.h>
#include <stdio.h>
static gboolean
cb_print_position (GstElement *pipeline)
{
gint64 pos;
if (gst_element_query_position (pipeline, GST_FORMAT_TIME, &pos))
{
g_print ("Time: %" GST_TIME_FORMAT "\r", GST_TIME_ARGS (pos));
}
/* call me again */
return TRUE;
}
static gboolean bus_call (GstBus *bus, GstMessage *msg, gpointer data)
{
GMainLoop *loop = (GMainLoop *) data;
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_EOS:
g_print ("End of stream\n");
g_main_loop_quit (loop);
break;
case GST_MESSAGE_ERROR: {
gchar *debug;
GError *error;
gst_message_parse_error (msg, &error, &debug);
g_free (debug);
g_printerr ("Error: %s\n", error->message);
g_error_free (error);
g_main_loop_quit (loop);
break;
}
default:
break;
}
return TRUE;
}
int main (int argc, char *argv[])
{
GMainLoop *loop;
GstElement *pipeline, *videotestsrcm, *x264encm, *rtmpsinkm, *flvmuxm;
GstBus *bus;
guint bus_watch_id;
/* Initialisation */
gst_init (&argc, &argv);
const gchar*nano_str;guint major, minor, micro, nano;
gst_init (&argc, &argv);
gst_version (&major, &minor, &micro, &nano);
if (nano == 1)
nano_str = "(CVS)";
else if (nano == 2)
nano_str = "(Prerelease)";
else
nano_str = "";
printf ("This program is linked against GStreamer %d.%d.%d%s\n",major, minor, micro, nano_str);
loop = g_main_loop_new (NULL, FALSE);
/* Create gstreamer elements */
pipeline = gst_pipeline_new ("videotest-pipeline");
videotestsrcm = gst_element_factory_make ("videotestsrc", "testsource");
x264encm = gst_element_factory_make ("x264enc", "videoencoder");
rtmpsinkm = gst_element_factory_make ("rtmpsink", "video2sink");
flvmuxm = gst_element_factory_make ("flvmux", "muxer");
if (!pipeline || !videotestsrcm || !x264encm || !rtmpsinkm || !flvmuxm) {
g_printerr ("One element could not be created. Exiting.\n");
return -1;
}
g_object_set (G_OBJECT (rtmpsinkm), "location" , argv[1] , NULL);
/* Set up the pipeline */
/* we add a message handler */
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
bus_watch_id = gst_bus_add_watch (bus, bus_call, loop);
gst_object_unref (bus);
/* we add all elements into the pipeline */
gst_bin_add_many (GST_BIN (pipeline),
videotestsrcm, x264encm, rtmpsinkm, flvmuxm, NULL);
/* we link the elements together */
/* videotestsrcm -> autovideosinkm */
gst_element_link_many (videotestsrcm, x264encm, flvmuxm, rtmpsinkm, NULL);
/* Set the pipeline to "playing" state*/
g_print ("Now set pipeline in state playing...\n");
gst_element_set_state (pipeline, GST_STATE_PLAYING);
/* run pipeline */
g_timeout_add (200, (GSourceFunc) cb_print_position, pipeline);
/* Iterate */
g_print ("Running...\n");
g_main_loop_run (loop);
/* Out of the main loop, clean up nicely */
g_print ("Returned, stopping playback\n");
gst_element_set_state (pipeline, GST_STATE_NULL);
g_print ("Deleting pipeline\n");
gst_object_unref (GST_OBJECT (pipeline));
g_source_remove (bus_watch_id);
g_main_loop_unref (loop);
return 0;
}

Gstreamer pipeline works with gst-launch but not in code. Reproducing a mjpeg stream from a IP camera

I want to reproduce a mjpeg stream from a intercom (but it's equivalent to a IP camera). Using gst-launch in the console works fine:
gst-launch-1.0 souphttpsrc location="http://192.168.1.191/api/camera/snapshot?width=640&height=480&fps=10" timeout=5 ! multipartdemux ! jpegdec ! videoconvert ! ximagesink
However, when I try to build an application to do this, it doesn't work.
My code:
#include <gst/gst.h>
#include <glib.h>
/* Structure to contain all our information, so we can pass it to callbacks */
typedef struct _CustomData {
GstElement *pipeline;
GstElement *source;
GstElement *v_demux;
GstElement *v_decoder;
GstElement *v_convert;
GstElement *v_sink;
} CustomData;
/* Handler for the pad-added signal */
static void pad_added_handler (GstElement *src, GstPad *pad, CustomData *data);
/** Main function */
int main(int argc, char *argv[]) {
CustomData data;
GstBus *bus;
GstMessage *msg;
GstStateChangeReturn ret;
gboolean terminate = FALSE;
/* Initialize GStreamer */
gst_init (&argc, &argv);
/* Create the elements
*
* souphttpsrc -> multipartdemux (~>) jpegdec -> videoconvert -> ximagesink
*
* ~> Sometimes pad
*
* */
data.source = gst_element_factory_make ("souphttpsrc", "video_source");
data.v_demux = gst_element_factory_make ("multipartdemux", "video_demux");
data.v_decoder = gst_element_factory_make ("jpegdec", "video_decoder");
data.v_convert = gst_element_factory_make ("videoconvert", "video_convert");
data.v_sink = gst_element_factory_make ("ximagesink", "video_sink");
/* Create the empty pipeline */
data.pipeline = gst_pipeline_new ("new-pipeline");
if (!data.pipeline || !data.source ||
!data.v_demux || !data.v_decoder || !data.v_convert || !data.v_sink ) {
g_printerr ("Not all elements could be created.\n");
return -1;
}
/* Configure elements */
g_object_set(G_OBJECT(data.source), "location", argv[1], NULL);
g_object_set(G_OBJECT(data.source), "timeout", 5, NULL);
/* Link all elements that can be automatically linked because they have "Always" pads */
gst_bin_add_many (GST_BIN (data.pipeline), data.source,
data.v_demux, data.v_decoder, data.v_convert, data.v_sink,
NULL);
if (gst_element_link_many (data.source, data.v_demux, NULL) != TRUE ||
gst_element_link_many (data.v_decoder, data.v_convert, data.v_sink, NULL) != TRUE ) {
g_printerr ("Elements could not be linked.\n");
gst_object_unref (data.pipeline);
return -1;
}
/* Connect to the pad-added signal */
g_signal_connect (data.v_demux, "pad-added", G_CALLBACK (pad_added_handler), &data);
/* Start playing */
ret = gst_element_set_state (data.pipeline, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE) {
g_printerr ("Unable to set the pipeline to the playing state.\n");
gst_object_unref (data.pipeline);
return -1;
}
/* Listen to the bus */
bus = gst_element_get_bus (data.pipeline);
do {
msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE,
GST_MESSAGE_STATE_CHANGED | GST_MESSAGE_ERROR | GST_MESSAGE_EOS);
/* Parse message */
if (msg != NULL) {
GError *err;
gchar *debug_info;
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_ERROR:
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);
terminate = TRUE;
break;
case GST_MESSAGE_EOS:
g_print ("End-Of-Stream reached.\n");
terminate = TRUE;
break;
case GST_MESSAGE_STATE_CHANGED:
/* We are only interested in state-changed messages from the pipeline */
if (GST_MESSAGE_SRC (msg) == GST_OBJECT (data.pipeline)) {
GstState old_state, new_state, pending_state;
gst_message_parse_state_changed (msg, &old_state, &new_state, &pending_state);
g_print ("Pipeline state changed from %s to %s:\n",
gst_element_state_get_name (old_state), gst_element_state_get_name (new_state));
}
break;
default:
/* We should not reach here */
g_printerr ("Unexpected message received.\n");
break;
}
gst_message_unref (msg);
}
} while (!terminate);
/* Free resources */
gst_object_unref (bus);
gst_element_set_state (data.pipeline, GST_STATE_NULL);
gst_object_unref (data.pipeline);
return 0;
}
/* This function will be called by the pad-added signal */
static void pad_added_handler (GstElement *src, GstPad *new_pad, CustomData *data) {
GstPad *sink_pad = NULL;
GstPadLinkReturn ret;
GstCaps *new_pad_caps = NULL;
GstStructure *new_pad_struct = NULL;
const gchar *new_pad_type = NULL;
g_print ("Received new pad '%s' from '%s':\n", GST_PAD_NAME (new_pad), GST_ELEMENT_NAME (src));
/* Get information of the new pad's type */
new_pad_caps = gst_pad_get_current_caps (new_pad);
new_pad_struct = gst_caps_get_structure (new_pad_caps, 0);
new_pad_type = gst_structure_get_name (new_pad_struct);
/* Get pad from the correspondent converter */
if (g_str_has_prefix (new_pad_type, "video")) {
sink_pad = gst_element_get_static_pad (data->v_decoder, "sink");
} else {
g_print (" It has type '%s' -> So exit\n", new_pad_type);
return;
}
/* If our converter is already linked, we have nothing to do here */
if (gst_pad_is_linked (sink_pad)) {
g_print (" We are already linked. Ignoring.\n");
gst_object_unref (sink_pad);
return;
}
ret = gst_pad_link (new_pad, sink_pad);
if (GST_PAD_LINK_FAILED (ret)) {
g_print (" Type is '%s' but link failed.\n", new_pad_type);
} else {
g_print (" Link succeeded (type '%s').\n", new_pad_type);
}
/* Unreference the new pad's caps, if we got them */
if (new_pad_caps != NULL) {
gst_caps_unref (new_pad_caps);
}
/* Unreference the sink pad */
if (sink_pad != NULL) {
gst_object_unref (sink_pad);
}
}
The output when I run the program:
Pipeline state changed from NULL to READY:
Pipeline state changed from READY to PAUSED:
Error received from element video_demux: Could not demultiplex stream.
Debugging information: multipartdemux.c(475): multipart_parse_header (): /GstPipeline:new-pipeline/GstMultipartDemux:video_demux:
Boundary not found in the multipart header
Any idea what am I missing?
Thanks in advance.
I have found sometimes that adding queues helps, maybe one before the jpegdec? Also maybe try a jpegparse before the jpegdec.

creating a pipeline to transmit voice

i have the following pipelines that one of them sends voice signals on udp port and the other receives them on the same port number on the receiver side
gst-launch-1.0 -v alsasrc ! audioconvert
! audio/x-raw,channels=2,depth=16,width=16,rate=44100 !
rtpL16pay ! udpsink
host=127.0.0.1 port=5000 //sender
and
gst-launch-1.0 udpsrc port=5000 ! "application/x-rtp,
media=(string)audio, clock-rate=(int)44100,
encoding-name=(string)L16, channels=(int)2,
payload=(int)96" ! rtpL16depay ! audioconvert
! alsasink //receiver
These pipelines work perfectly.
now i am trying to write a source code using Gstreamer SDK that does the same thing. I have come so far:
#include <gst/gst.h>
#include <string.h>
int main(int argc, char *argv[]) {
GstElement *pipeline, *source, *audiosink,*rtppay,*rtpdepay,*filter,*filter1,*conv,*conv1,*udpsink,*udpsrc,*receive_resample;
GstBus *bus;
GstMessage *msg;
GstCaps *filtercaps;
GstStateChangeReturn ret;
/* Initialize GStreamer */
gst_init (&argc, &argv);
/* Create the elements */
source = gst_element_factory_make ("alsasrc", "source");
conv= gst_element_factory_make ("audioconvert", "conv");
conv1= gst_element_factory_make ("audioconvert", "conv1");
filter=gst_element_factory_make("capsfilter","filter");
rtppay=gst_element_factory_make("rtpL16pay","rtppay");
rtpdepay=gst_element_factory_make("rtpL16depay","rtpdepay");
udpsink=gst_element_factory_make("udpsink","udpsink");
audiosink = gst_element_factory_make ("autoaudiosink", "audiosink");
receive_resample = gst_element_factory_make("audioresample", NULL);
udpsrc=gst_element_factory_make("udpsrc",NULL);
filter1=gst_element_factory_make("capsfilter","filter");
g_object_set(udpsrc,"port",5000,NULL);
g_object_set (G_OBJECT (udpsrc), "caps", gst_caps_from_string("application/x-rtp,media=audio,payload=96,clock-rate=44100,encoding-name=L16,channels=2"), NULL);
/* Create the empty pipeline */
pipeline = gst_pipeline_new ("test-pipeline");
if (!pipeline || !source || !filter || !conv || !rtppay || !udpsink ) {
g_printerr ("Not all elements could be created.\n");
return -1;
}
g_object_set(G_OBJECT(udpsink),"host","127.0.0.1",NULL);
g_object_set(G_OBJECT(udpsink),"port",5000,NULL);
filtercaps = gst_caps_new_simple ("audio/x-raw",
"channels", G_TYPE_INT, 2,
"width", G_TYPE_INT, 16,
"depth", G_TYPE_INT, 16,
"rate", G_TYPE_INT, 44100,
NULL);
g_object_set (G_OBJECT (filter), "caps", filtercaps, NULL);
gst_caps_unref (filtercaps);
filtercaps = gst_caps_new_simple ("application/x-rtp",
"media",G_TYPE_STRING,"audio",
"clock-rate",G_TYPE_INT,44100,
"encoding-name",G_TYPE_STRING,"L16",
"channels", G_TYPE_INT, 2,
"payload",G_TYPE_INT,96,
NULL);
g_object_set (G_OBJECT (filter1), "caps", filtercaps, NULL);
gst_caps_unref (filtercaps);
/* Build the pipeline */
gst_bin_add_many (GST_BIN (pipeline), source,filter,conv,rtppay,udpsink, NULL);
if (gst_element_link_many (source,filter,conv,rtppay,udpsink, NULL) != TRUE) {
g_printerr ("Elements could not be linked.\n");
gst_object_unref (pipeline);
return -1;
}
gst_bin_add_many (GST_BIN (pipeline),udpsrc,rtpdepay,conv1,receive_resample,audiosink,NULL);
if (gst_element_link_many (udpsrc,rtpdepay,conv1,receive_resample,audiosink,NULL) != TRUE) {
g_printerr ("Elements could not be linked.\n");
gst_object_unref (pipeline);
return -1;
}
/* Modify the source's properties */
// g_object_set (source, "pattern", 0, NULL);
/* Start playing */
ret = gst_element_set_state (pipeline, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE) {
g_printerr ("Unable to set the pipeline to the playing state.\n");
gst_object_unref (pipeline);
return -1;
}
/* Wait until error or EOS */
bus = gst_element_get_bus (pipeline);
msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE, GST_MESSAGE_ERROR | GST_MESSAGE_EOS);
/* Parse message */
if (msg != NULL) {
GError *err;
gchar *debug_info;
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_ERROR:
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);
break;
case GST_MESSAGE_EOS:
g_print ("End-Of-Stream reached.\n");
break;
default:
/* We should not reach here because we only asked for ERRORs and EOS */
g_printerr ("Unexpected message received.\n");
break;
}
gst_message_unref (msg);
}
/* Free resources */
gst_object_unref (bus);
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (pipeline);
return 0;
}
but somehow i dont receive any voice on the receiver. i dont get any errors of any kind. Any ideas why this is happening?

RTSP pipeline implemented via C code not working?

My Scenario is as follows :-
I have set up a RTSP server at IP 192.168.1.24 at port 554.I use the following gst-launch command on client side to receive packets and everything works fine.
gst-launch rtspsrc location = rtsp://admin:admin123#192.168.1.24:554/axis-media/media.amp ! fakesink
But when I implement the same thing via C code it gives me error.My C code is as follows:-
#include <gst.h>
#include <glib.h>
static gboolean bus-call (GstBus *bus, GstMessage *msg, gpointer data)
{
GMainLoop *loop = (GMainLoop *) data;
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_EOS:
g_print ("End of stream\n");
g_main_loop_quit (loop);
break;
case GST_MESSAGE_ERROR: {
gchar *debug;
GError *error;
gst_message_parse_error (msg, &error, &debug);
g_free (debug);
g_printerr ("Error: %s\n", error->message);
g_error_free (error);
g_main_loop_quit (loop);
break;
}
default:
break;
}
return true;
}
int main (int argc, char *argv[])
{
GMainLoop *loop;
GstElement *pipeline, *source, *sink;
GstBus *bus;
gst_init (&argc, &argv);
loop = g_main_loop_new (NULL, FALSE);
if (argc != 2) {
return -1;
}
pipeline = gst_pipeline_new ("network-player");
source = gst_element_factory_make ("rtspsrc","file-source");
sink = gst_element_factory_make ("fakesink","fake");
if (!pipeline || !source || !sink) {
g_printerr ("One element could not be created. Exiting.\n");
return -1;
}
g_object_set (G_OBJECT (source), "location", argv[1], NULL);
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
gst_bus_add_watch (bus, bus_call, loop);
gst_object_unref (bus);
gst_bin_add_many (GST_BIN (pipeline),source, sink, NULL);
gst_element_link_many (source, sink, NULL);
/* Set the pipeline to "playing" state*/
gst_element_set_state (pipeline, GST_STATE_PLAYING);
/* Iterate */
g_print ("Running...\n");
g_main_loop_run (loop);
/* Out of the main loop, clean up nicely */
g_print ("Returned, stopping playback\n");
gst_element_set_state (pipeline, GST_STATE_NULL);
g_print ("Deleting pipeline\n");
gst_object_unref (GST_OBJECT (pipeline));
return 0;
}
I am able to compile the code without any error.
But when I run the binary generated with the following format:-
user#user:~ ./helloworld rtsp://admin:admin123#192.168.1.24:554/axis-media/media.amp
I get the following error:-
Now playing: rtsp://root:nlss123#192.168.1.24:554/axis-media/media.amp
Running...
**Error: Internal data flow error**.
Returned, stopping playback
Deleting pipeline
Can anyone suggest we there is Internal Data flow error ?
i also had the same problem.
You should link source to to sink with "pad-added" signal.
In brief:
typedef struct myDataTag {
GstElement *pipeline;
GstElement *rtspsrc;
GstElement *depayloader;
GstElement *decoder;
*sink;
} myData_t;
myData_t appData;
appData->pipeline = gst_pipeline_new ("videoclient");
appData->rtspsrc = gst_element_factory_make ("rtspsrc", "rtspsrc");
g_object_set (G_OBJECT (appData->rtspsrc), "location", "rtsp://192.168.1.10:554/myStreamPath", NULL);
appData->depayloader = gst_element_factory_make ("rtph264depay","depayloader");
appData->decoder = gst_element_factory_make ("h264dec", "decoder");
appData->sink = gst_element_factory_make ("autovideosink", "sink");
//then add all elements together
gst_bin_add_many (GST_BIN (appData->pipeline), appData->rtspsrc, appData->depayloader, appData->decoder, appData->sink, NULL);
//link everythink after source
gst_element_link_many (appData->depayloader, appData->decoder, appData->sink, NULL);
/*
* Connect to the pad-added signal for the rtpbin. This allows us to link
* the dynamic RTP source pad to the depayloader when it is created.
*/
g_signal_connect (appData->rtspsrc, "pad-added", G_CALLBACK (pad_added_handler), &appData);
/* Set the pipeline to "playing" state*/
gst_element_set_state (appData->pipeline, GST_STATE_PLAYING);
/* pad added handler */
static void pad_added_handler (GstElement *src, GstPad *new_pad, myData_t *pThis) {
GstPad *sink_pad = gst_element_get_static_pad (pThis->depayloader, "sink");
GstPadLinkReturn ret;
GstCaps *new_pad_caps = NULL;
GstStructure *new_pad_struct = NULL;
const gchar *new_pad_type = NULL;
g_print ("Received new pad '%s' from '%s':\n", GST_PAD_NAME (new_pad), GST_ELEMENT_NAME (src));
/* Check the new pad's name */
if (!g_str_has_prefix (GST_PAD_NAME (new_pad), "recv_rtp_src_")) {
g_print (" It is not the right pad. Need recv_rtp_src_. Ignoring.\n");
goto exit;
}
/* If our converter is already linked, we have nothing to do here */
if (gst_pad_is_linked (sink_pad)) {
g_print (" Sink pad from %s already linked. Ignoring.\n", GST_ELEMENT_NAME (src));
goto exit;
}
/* Check the new pad's type */
new_pad_caps = gst_pad_get_caps (new_pad);
new_pad_struct = gst_caps_get_structure (new_pad_caps, 0);
new_pad_type = gst_structure_get_name (new_pad_struct);
/* Attempt the link */
ret = gst_pad_link (new_pad, sink_pad);
if (GST_PAD_LINK_FAILED (ret)) {
g_print (" Type is '%s' but link failed.\n", new_pad_type);
} else {
g_print (" Link succeeded (type '%s').\n", new_pad_type);
}
exit:
/* Unreference the new pad's caps, if we got them */
if (new_pad_caps != NULL)
gst_caps_unref (new_pad_caps);
/* Unreference the sink pad */
gst_object_unref (sink_pad);
}
Hope that this will help someone..:)
you can get verbose error logs by running the apps by --gst-debug=*rtsp*:5 e.g.
./yourApplication --gst-debug=*rtsp*:5

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