Pushing images into a gstreamer pipeline - c++

After playing around with some toy applications, exploring the
documentation and googling around (including the mailing list
archives) I am still puzzled for what I would think is a rather common
use case.
I have an existing code that generates images (in memory) and I would
like to push these images into a gstreamer pipeline (to create a flv
video at the end).
I could not find an "obvious way to do it". My best guess will be to
dig in the source code of GstMultiFileSrc and its parent GstPushSrc,
to figure it out.
Could any of you point me out to the "obvious way" of doing this ?
Is it there any related piece of documentation/tutorial/example on this ?
Once I have the input right, the rest is a piece of cake, thanks to
Gstreamer awesomeness !
(something like "my magic input -> ffmpegcolorspace ! ffenc_flv !
flvmux ! filesink location=desktop.flv" )
Thanks for your answers.

GStreamer uses plugins to do everything. Plugins that create data or take it from an external source are called "src" plugins.
The generic src plugin for injecting application-generated data into a pipeline is called appsrc. The API provided by appsrc is documented as part of the App Library.
Here's one example that demonstrates feeding appsrc with generated images: gdk-gstappsrc-stream.c. It seems to be derived from some test code in the GStreamer source tree: here.
Another approach would be to create your own src plugin. Look at the goom music visualization plugin for an example that seems to work in a way similar to what you have specified.

I found a solution (maybe) to this (i get the images with OpenCV) ... but i have an error with the pipeline: ERROR from element mysource: Error en el flujo de datos interno.
Debugging info: gstbasesrc.c(2574): gst_base_src_loop (): /GstPipeline:pipeline0/GstAppSrc:mysource:
streaming task paused, reason not-negotiated (-4)
this is the code:
typedef struct _App App;
struct _App{
GstElement *pipeline;
GstElement *appsrc;
GMainLoop *loop;
guint sourceid;
GTimer *timer;
};
App s_app;
CvCapture *capture;
static gboolean read_data(App *app){
GstFlowReturn ret;
GstBuffer *buffer = gst_buffer_new();
IplImage* frame = cvQueryFrame(capture);
GST_BUFFER_DATA(buffer) = (uchar*)frame->imageData;
GST_BUFFER_SIZE(buffer) = frame->width*frame->height*sizeof(uchar*);
g_signal_emit_by_name(app->appsrc,"push-buffer",buffer,&ret);
gst_buffer_unref(buffer);
if(ret != GST_FLOW_OK){
GST_DEBUG("Error al alimentar buffer");
return FALSE;
}
return TRUE;
}
static void start_feed(GstElement* pipeline,guint size, App* app){
if(app->sourceid == 0){
GST_DEBUG("Alimentando");
app->sourceid = g_idle_add((GSourceFunc) read_data, app);
}
}
static void stop_feed(GstElement* pipeline, App* app){
if(app->sourceid !=0 ){
GST_DEBUG("Stop feeding");
g_source_remove(app->sourceid);
app->sourceid = 0;
}
}
static gboolean
bus_message (GstBus * bus, GstMessage * message, App * app)
{
GST_DEBUG ("got message %s",
gst_message_type_get_name (GST_MESSAGE_TYPE (message)));
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_ERROR: {
GError *err = NULL;
gchar *dbg_info = NULL;
gst_message_parse_error (message, &err, &dbg_info);
g_printerr ("ERROR from element %s: %s\n",
GST_OBJECT_NAME (message->src), err->message);
g_printerr ("Debugging info: %s\n", (dbg_info) ? dbg_info : "none");
g_error_free (err);
g_free (dbg_info);
g_main_loop_quit (app->loop);
break;
}
case GST_MESSAGE_EOS:
g_main_loop_quit (app->loop);
break;
default:
break;
}
return TRUE;
}
int main(int argc, char* argv[]){
App *app = &s_app;
GError *error = NULL;
GstBus *bus;
GstCaps *caps;
capture = cvCaptureFromCAM(0);
gst_init(&argc,&argv);
/* create a mainloop to get messages and to handle the idle handler that will
* feed data to appsrc. */
app->loop = g_main_loop_new (NULL, TRUE);
app->timer = g_timer_new();
app->pipeline = gst_parse_launch("appsrc name=mysource ! video/x-raw-rgb,width=640,height=480,bpp=24,depth=24 ! ffmpegcolorspace ! videoscale method=1 ! theoraenc bitrate=150 ! tcpserversink host=127.0.0.1 port=5000", NULL);
g_assert (app->pipeline);
bus = gst_pipeline_get_bus (GST_PIPELINE (app->pipeline));
g_assert(bus);
/* add watch for messages */
gst_bus_add_watch (bus, (GstBusFunc) bus_message, app);
/* get the appsrc */
app->appsrc = gst_bin_get_by_name (GST_BIN(app->pipeline), "mysource");
g_assert(app->appsrc);
g_assert(GST_IS_APP_SRC(app->appsrc));
g_signal_connect (app->appsrc, "need-data", G_CALLBACK (start_feed), app);
g_signal_connect (app->appsrc, "enough-data", G_CALLBACK (stop_feed), app);
/* set the caps on the source */
caps = gst_caps_new_simple ("video/x-raw-rgb",
"bpp",G_TYPE_INT,24,
"depth",G_TYPE_INT,24,
"width", G_TYPE_INT, 640,
"height", G_TYPE_INT, 480,
NULL);
gst_app_src_set_caps(GST_APP_SRC(app->appsrc), caps);
/* go to playing and wait in a mainloop. */
gst_element_set_state (app->pipeline, GST_STATE_PLAYING);
/* this mainloop is stopped when we receive an error or EOS */
g_main_loop_run (app->loop);
GST_DEBUG ("stopping");
gst_element_set_state (app->pipeline, GST_STATE_NULL);
gst_object_unref (bus);
g_main_loop_unref (app->loop);
cvReleaseCapture(&capture);
return 0;
}
Any idea???

You might try hacking imagefreeze to do what you want. appsrc might also do it.

Related

Why the gstreamer pipeline won't play if I added a udpsink plugin to it

I tried to send an RTP stream with gstreamer, but I found that the pipeline won't play at all. When I simplified my code I found that if the udpsink plugin was added in the pipeline, the pipeline is blocked, and the status is always READY.
My code:
#include <gst/gst.h>
int main(int argc, char *argv[]) {
GstElement *pipeline, *source, *sink, *udp, *convert;
GstBus *bus;
GstMessage *msg;
GstStateChangeReturn ret;
gboolean terminate = FALSE;
/* Initialize GStreamer */
gst_init (&argc, &argv);
/* Create the elements */
source = gst_element_factory_make ("videotestsrc", "source");
convert = gst_element_factory_make ("videoconvert", "convert");
sink = gst_element_factory_make ("autovideosink", "sink");
udp = gst_element_factory_make ("udpsink", "udp");
/* Create the empty pipeline */
pipeline = gst_pipeline_new ("test-pipeline");
/* Build the pipeline */
gst_bin_add_many (GST_BIN (pipeline), source, sink, convert, /*udp,*/ NULL);
gst_element_link_many (source, convert, sink, NULL);
/* 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;
}
bus = gst_element_get_bus (pipeline);
do {
msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE, GST_MESSAGE_ERROR | GST_MESSAGE_STATE_CHANGED | GST_MESSAGE_EOS);
/* Parse message */
if (msg != NULL) {
GError *err;
gchar *debug_info;
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_ERROR:
case GST_MESSAGE_EOS:
terminate = TRUE;
break;
case GST_MESSAGE_STATE_CHANGED:
if (GST_MESSAGE_SRC (msg) == GST_OBJECT (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 because we only asked for ERRORs and EOS */
g_printerr ("Unexpected message received.\n");
break;
}
gst_message_unref (msg);
}
} while (!terminate);
/* Free resources */
// ...
}
As you can see, the pipeline works fine if the udpsink is not added. This also happens in command line:
gst-launch-1.0 -v udpsink videotestsrc ! videoconvert ! autovideosink
The command above will popup a window and the video stops at the first frame.
I don't know what's wrong with my code, can anyone give me help, thanks!
A pipeline ends with a sink element, e.g. (autovideosink) you cannot add anything after that. That is why autovideosink does not have src pad and you can not link it to udpsink. You have to have a second thread linked to udpsink. To create threads you can use queue and tee elements.
You can find more here (GStreamer Basic tutorial 7: Multithreading and Pad Availability)

pad creation and some mistakes on gst_parse_launch

I am a newbie to gstreamer and I would like to know if we have to create source and sink pads for convert like video convert in a pipeline. I have a pipeline like this
gst-launch-1.0 v4l2src ! video/x-raw,format=YUY2 ! videoconvert ! xvimagesink
I am trying to create a simple c application to understand the creation of pads and would like to know if video convert has a source pad and sink pad too. I am creating a source and sink pad for the filter.
EDIT:
yeah well you see, I tried following the dynamic pipelines example and wrote the code below
#include <gst/gst.h>
// easier to pass them as callbacks
typedef struct _CustomData{
GstElement *pipeline;
GstElement *source;
GstElement *convert;
GstElement *sink;
}CustomData;
// callback function
// here src is the v4l2src, newpad is gstpad that has just been added to src element. This is usually the pad to which we want to lnk
// data is the pointer we provided when attaching to the signal.
static void pad_added_handler(GstElement *src, GstPad *new_pad,CustomData *data)
{
g_print("In pad handler\n");
GstPad *sink_pad = gst_element_get_static_pad(data->convert, "sink");
GstPadLinkReturn ret;
GstCaps *new_pad_caps = NULL;
GstStructure *new_pad_struct = NULL;
const gchar *new_pad_type = NULL;
if(gst_pad_is_linked(sink_pad))
{
g_print("we are linked. igonring\n");
}
// check the new pad types
// we have previously created a piece of pipeline which deals with videoconvert linked with xvimagesink and we will nto be able to link it to a pad producing video.
//gst-pad_get_current_caps()- retrieves current capabilities of pad
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);
g_print ("It has type '%s' which is not raw audio. Ignoring.\n", new_pad_type);
if(!g_str_has_prefix(new_pad_type, "video/x-raw,format=(string)YUY2"))
{
g_print("It has new pad type");
}
// gst_pad_link tries to link two pads . the link must be specified from source to sink and both pads must be owned by elements residing in same pipeline
ret = gst_pad_link(new_pad, sink_pad);
if(GST_PAD_LINK_FAILED(ret))
{
g_print("type is new_pad_type");
}
if(new_pad_caps !=NULL)
{
gst_caps_unref(new_pad_caps);
}
gst_object_unref(sink_pad);
}
int main(int argc, char *argv[])
{
GMainLoop *loop;
CustomData data;
GstBus *bus;
GstMessage *msg;
gboolean terminate = FALSE;
gst_init(&argc, &argv);
// loop = g_main_loop_new(NULL, FALSE);
// create the elements
data.source = gst_element_factory_make("v4l2src", "source");
data.convert = gst_element_factory_make("videoconvert", "convert");
data.sink = gst_element_factory_make("xvimagesink", "sink");
data.pipeline = gst_pipeline_new("new-pipeline");
if(!data.pipeline || !data.source || !data.convert || !data.sink)
{
g_printerr("Not all elements could be created\n");
return -1;
}
//we did not link source at this point of time, we will do it later
gst_bin_add_many(GST_BIN(data.pipeline), data.source, data.convert, data.sink, NULL);
// we link convert element to sink, do not link them with source. we dont have source pads here. so we just have videoconvert->sink unlinked
// gst_element_link(data.source, data.convert);
if(!gst_element_link( data.convert,data.sink))
{
g_printerr("elements could not be linked\n");
gst_object_unref(data.pipeline);
return -1;
}
// we set the device source
//g_object_set(source, "device", "/dev/video0", NULL);
//connect to pad added signal.
// we want to attach pad added signal to source element. to do so, we are using g_signal_connect and provide callback function and datapointer.
// when source element has enough information to start producing data, it will create source pads and trigger the pad added signal. at this point, our callback is called
g_print("before signal connect\n");
gint id= g_signal_connect(G_OBJECT(data.source), "pad-added", G_CALLBACK(pad_added_handler), &data );
g_print("after signal connect with id = %d\n", id);
//g_signal_connect(G_OBJECT(data.source), "pad-added", G_CALLBACK(handler), &data);
// gst_element_link(data.source, data.convert);
GstStateChangeReturn ret;
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;
}
// g_main_loop_run(loop);
/* 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;
}
and it gave me an error
before signal connect
after signal connect with id = 1
Pipeline state changed from NULL to READY:
Pipeline state changed from READY to PAUSED:
Error received from element source: Internal data stream error.
Debugging information: gstbasesrc.c(3055): gst_base_src_loop (): /GstPipeline:new-pipeline/GstV4l2Src:source:
streaming stopped, reason not-linked (-1)
(The above code works if I write gst_elements_link(data.source, data.convert) after the element link statement for convert and sink)
So I tried the normal way in which I just added and linked all the elements together and it began to work without use of the pads.
#include <gst/gst.h>
int main(int argc, char *argv[])
{
GstElement *pipeline, *source, *convert, *sink;
GstBus *bus;
GstMessage *msg;
gst_init(&argc, &argv);
source = gst_element_factory_make("v4l2src", "source");
convert = gst_element_factory_make("nvvidconv", "convert");
sink = gst_element_factory_make("xvimagesink", "sink");
pipeline = gst_pipeline_new("pipe");
gst_bin_add_many(GST_BIN(pipeline), source,convert,sink, NULL);
gst_element_link_many(source,convert,sink,NULL);
gst_element_set_state(pipeline,GST_STATE_PLAYING);
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);
}
But, inorder to fully grasp the knowledge of pads, I wanted to execute simpler pipelines with pads.
I just don't fully understand the usage of pads and link them and all.
EDIT2:
Ultimately I want to write application for pipeline like this which works on command line perfectly well,
gst-launch-1.0 v4l2src device='/dev/video0' ! 'video/x-raw,format=(string)YUY2,width=(int)640,height=(int)480' ! nvvidconv ! 'video/x-raw(memory:NVMM),format=(string)NV12,width=(int)640,height=(int)480' ! nvvidconv ! 'video/x-raw,format=(string)NV12,width=(int)640,height=(int)480' ! nvvideoconvert ! 'video/x-raw(memory:NVMM),format=(string)NV12,width=(int)640,height=(int)480' ! mux.sink_0 nvstreammux live-source=1 name=mux batch-size=1 width=640 height=480 ! nvinfer config-file-path=/opt/nvidia/deepstream/deepstream-4.0/sources/apps/sample_apps/deepstream-test1/dstest1_pgie_config.txt batch-size=1 ! nvmultistreamtiler rows=1 columns=1 width=640 height=480 ! nvvideoconvert ! nvdsosd ! nvegltransform ! nveglglessink sync=false -v
But as I don't understand the usage of both pads and bins, I am unable to implement them in my above pipeline. However, I tried this,
#include <gst/gst.h>
#include <glib.h>
#include <stdio.h>
#include "gstnvdsmeta.h"
#define MAX_DISPLAY_LEN 64
#define PGIE_CLASS_ID_VEHICLE 0
#define PGIE_CLASS_ID_PERSON 2
gint frame_number = 0;
gchar pgie_classes_str[4][32] = { "Vehicle", "TwoWheeler", "Person",
"Roadsign"
};
static GstPadProbeReturn osd_sink_pad_buffer_probe(GstPad *pad, GstPadProbeInfo *info, gpointer u_data)
{
GstBuffer *buf=(GstBuffer *)info->data;
guint num_rects =0;
NvDsObjectMeta *obj_meta = NULL;
guint vehicle_count = 0;
guint person_count = 0;
NvDsMetaList * l_frame = NULL;
NvDsMetaList * l_obj = NULL;
NvDsDisplayMeta *display_meta = NULL;
NvDsBatchMeta *batch_meta = gst_buffer_get_nvds_batch_meta (buf);
for (l_frame = batch_meta->frame_meta_list; l_frame != NULL;
l_frame = l_frame->next) {
NvDsFrameMeta *frame_meta = (NvDsFrameMeta *) (l_frame->data);
int offset = 0;
for (l_obj = frame_meta->obj_meta_list; l_obj != NULL;
l_obj = l_obj->next) {
obj_meta = (NvDsObjectMeta *) (l_obj->data);
if (obj_meta->class_id == PGIE_CLASS_ID_VEHICLE) {
vehicle_count++;
num_rects++;
}
if (obj_meta->class_id == PGIE_CLASS_ID_PERSON) {
person_count++;
num_rects++;
}
}
display_meta = nvds_acquire_display_meta_from_pool(batch_meta);
NvOSD_TextParams *txt_params = &display_meta->text_params[0];
display_meta->num_labels = 1;
txt_params->display_text = g_malloc0 (MAX_DISPLAY_LEN);
offset = snprintf(txt_params->display_text, MAX_DISPLAY_LEN, "Person = %d ", person_count);
offset = snprintf(txt_params->display_text + offset , MAX_DISPLAY_LEN, "Vehicle = %d ", vehicle_count);
/* Now set the offsets where the string should appear */
txt_params->x_offset = 10;
txt_params->y_offset = 12;
/* Font , font-color and font-size */
txt_params->font_params.font_name = "Serif";
txt_params->font_params.font_size = 10;
txt_params->font_params.font_color.red = 1.0;
txt_params->font_params.font_color.green = 1.0;
txt_params->font_params.font_color.blue = 1.0;
txt_params->font_params.font_color.alpha = 1.0;
/* Text background color */
txt_params->set_bg_clr = 1;
txt_params->text_bg_clr.red = 0.0;
txt_params->text_bg_clr.green = 0.0;
txt_params->text_bg_clr.blue = 0.0;
txt_params->text_bg_clr.alpha = 1.0;
nvds_add_display_meta_to_frame(frame_meta, display_meta);
}
g_print ("Frame Number = %d Number of objects = %d "
"Vehicle Count = %d Person Count = %d\n",
frame_number, num_rects, vehicle_count, person_count);
frame_number++;
return GST_PAD_PROBE_OK;
}
int main(int argc, char *argv[])
{
GstElement *pipeline, *source, *filter1, *convert,*filter2, *filter3, *vidconv, *filter4, *mux, *infer, *tiler, *osd, *transform , *sink, *bin, *convert2 , *vidconv2;
GMainLoop *loop;
GstCaps *caps1, *caps2, *caps3, *caps4;
GstPad *osd_sink_pad =NULL, *srcpad, *sinkpad;
loop = g_main_loop_new(NULL,FALSE);
gst_init(&argc, &argv);
pipeline = gst_pipeline_new("nv_pipeline");
gchar *string1 = "video/x-raw(memory:NVMM),format=(string)NV12";
source = gst_element_factory_make("v4l2src", "source");
filter1 = gst_element_factory_make("capsfilter", "filter1");
convert = gst_element_factory_make("nvvidconv", "convert");
filter2 = gst_element_factory_make("capsfilter", "filter2");
filter3 = gst_element_factory_make("capsfilter", "filter3");
filter4 = gst_element_factory_make("capsfilter", "filter4");
vidconv = gst_element_factory_make("nvvideoconvert", "vidconv");
mux = gst_element_factory_make("nvstreammux", "mux");
infer = gst_element_factory_make("nvinfer", "infer");
tiler = gst_element_factory_make("nvmultistreamtiler", "tiler");
osd = gst_element_factory_make("nvosd", "osd");
transform = gst_element_factory_make("nvegltransform", "transform");
sink = gst_element_factory_make("nveglglessink", "sink");
convert2 = gst_element_factory_make("nvvidconv", "convert2");
vidconv2 = gst_element_factory_make("nvvideoconvert", "vidconv2");
gst_bin_add_many(GST_BIN(pipeline), source,filter1,convert,filter2, convert2,filter3,vidconv, filter4,mux,infer, tiler,vidconv2, osd,transform,sink,NULL);
gst_element_link_many(source,filter1,convert,filter2, convert2,filter3, vidconv, filter4,mux,infer, tiler,vidconv2, osd,transform,sink,NULL);
osd_sink_pad = gst_element_get_static_pad(osd, "sink");
gst_pad_add_probe(osd_sink_pad, GST_PAD_PROBE_TYPE_BUFFER, osd_sink_pad_buffer_probe, NULL, NULL);
caps1 = gst_caps_new_simple("video/x-raw", "format",G_TYPE_STRING,"YUY2",NULL);
caps2 = gst_caps_from_string(string1);
caps3 = gst_caps_new_simple("video/x-raw", "format", G_TYPE_STRING,"NV12", NULL);
caps4 = gst_caps_from_string(string1);
g_object_set(G_OBJECT(filter1), "caps", caps1, NULL);
g_object_set(G_OBJECT(filter2), "caps", caps2, NULL);
g_object_set(G_OBJECT(filter3), "caps", caps3, NULL);
g_object_set(G_OBJECT(filter4), "caps", caps4, NULL);
g_object_set(G_OBJECT(mux), "live-source", 1, "name", "mux", "batch-size", 1, "width", 1280, "height", 720, NULL);
g_object_set(G_OBJECT(infer), "config-file-path","/opt/nvidia/deepstream/deepstream-4.0/sources/apps/sample_apps/deepstream-test1/dstest1_pgie_config.txt",NULL);
g_object_set(G_OBJECT(infer), "batch-size", 1, NULL);
g_object_set(G_OBJECT(tiler), "rows", 1, "columns", 1, "width", 1280, "height", 720, NULL);
gst_caps_unref(caps1);
gst_caps_unref(caps2);
gst_caps_unref(caps3);
gst_caps_unref(caps4);
gst_element_set_state(pipeline, GST_STATE_PLAYING);
g_print("Running ...\n");
g_main_loop_run(loop);
gst_element_set_state(pipeline,GST_STATE_NULL);
gst_object_unref(pipeline);
return 0;
}
which gives the exact same output as the command line gst-launch-1.0 like,
(deep_pads:15648): GLib-GObject-WARNING **: 11:29:18.761: cannot register existing type 'GstInterpolationMethod'
(deep_pads:15648): GLib-GObject-CRITICAL **: 11:29:18.761: g_param_spec_enum: assertion 'G_TYPE_IS_ENUM (enum_type)' failed
(deep_pads:15648): GLib-GObject-CRITICAL **: 11:29:18.761: validate_pspec_to_install: assertion 'G_IS_PARAM_SPEC (pspec)' failed
(deep_pads:15648): GStreamer-CRITICAL **: 11:29:18.814: gst_element_get_static_pad: assertion 'GST_IS_ELEMENT (element)' failed
(deep_pads:15648): GStreamer-CRITICAL **: 11:29:18.814: gst_pad_add_probe: assertion 'GST_IS_PAD (pad)' failed
0:00:00.843318172 15648 0x5591be52c0 INFO nvinfer gstnvinfer.cpp:519:gst_nvinfer_logger:<infer> NvDsInferContext[UID 1]:initialize(): Trying to create engine from model files
0:00:20.693301580 15648 0x5591be52c0 INFO nvinfer gstnvinfer.cpp:519:gst_nvinfer_logger:<infer> NvDsInferContext[UID 1]:generateTRTModel(): Storing the serialized cuda engine to file at /opt/nvidia/deepstream/deepstream-4.0/samples/models/Primary_Detector/resnet10.caffemodel_b1_int8.engine
except It doesn't display an output window from the above c application and doesn't display the rest.
Your first example fails because data.source element is actually never linked to data.convert element. Since both elements have static pads you need to "manually" create them and link them before setting pipeline to GST_STATE_PLAYING:
GstPad *source_pad = gst_element_get_static_pad(data.source, "src");
GstPad *sink_pad = gst_element_get_static_pad(data.convert, "sink");
ret = gst_pad_link(source_pad, sink_pad);
You probably expected that static source pad of your data.source element will somehow be automatically created so you registered
g_signal_connect(G_OBJECT(data.source), "pad-added", G_CALLBACK(pad_added_handler), &data );
But, as you can see from your debug, pad_added_handler was never called because g_signal_connect can be registered and will be called for elements that have dynamic pads. For example, demultiplexer tsdemux will dynamically create its source pads during discovery of elementary streams so, in that case, registration of pad-added callback would be necessary.
The first step in your learning curve would be to understand fundamental differences between "static" (mandatory, always exists, "manually" created), dynamic (exists sometimes, automatically created by element) and request (optional, "manually" created when needed) gstreamer pads. After that everything will be much easier for you.

How to listen to oggdemux gstreamer failures?

I wrote a gstreamer app to convert from opus audio to raw audio. If I feed bad audio (just random bytes) to the pipeline, the pipeline gets stuck and /i don't receive an error message on the message bus.
I'm listening to the error messages flowing through the pipeline, but not getting an error code to indicate the failure. The gstreamer debug logs indicate the demux failed though, I can see the following in the logs:
0:00:00.021614679 22541 0xe5b190 WARN oggdemux gstoggdemux.c:4609:gst_ogg_demux_send_event:<oggdemux0> No chain to forward event to
0:00:00.021656681 22541 0xe5b190 WARN oggdemux gstoggdemux.c:2433:gst_ogg_demux_sink_event:<oggdemux0> EOS while trying to retrieve chain, seeking disabled
The following is an app sample that I wrote:
#include <gst/gst.h>
#include <gst/gstbin.h>
#include <gst/app/gstappsrc.h>
#include <gst/app/gstappsink.h>
#include <stdio.h>
#include <string.h>
static GMainLoop *loop;
FILE *file = NULL;
size_t bytesRead = 0;
typedef struct _CustomData
{
GstElement *pipeline;
GstAppSrc *app_source;
guint sourceid; /* To control the GSource */
} CustomData;
static gboolean push_data(CustomData *data)
{
GstBuffer *gbuffer;
GstFlowReturn ret;
char buffer[1024];
gbuffer = gst_buffer_new_and_alloc(sizeof(buffer));
GstMapInfo info;
bytesRead = fread(buffer, 1, sizeof(buffer), file);
gst_buffer_map(gbuffer, &info, GST_MAP_WRITE);
memcpy(info.data, buffer, bytesRead);
gst_buffer_unmap(gbuffer, &info);
if (bytesRead > 0)
{
//g_print("Pushing %d\n", (int)bytesRead);
/* Push the buffer into the appsrc */
g_signal_emit_by_name(data->app_source, "push-buffer", gbuffer, &ret);
return TRUE;
}
else
{
g_print("file complete\n");
gst_app_src_end_of_stream(data->app_source);
return FALSE;
}
gst_buffer_unref(gbuffer);
}
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;
}
}
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);
}
}
static gboolean bus_call(GstBus * bus, GstMessage * msg, gpointer user_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: from %s %s\n", GST_OBJECT_NAME(msg->src), error->message);
g_error_free(error);
g_main_loop_quit(loop);
break;
}
default:
break;
}
return TRUE;
}
int main(int argc,
char *argv[])
{
CustomData data;
memset(&data, 0, sizeof(data));
GstBus *bus;
guint bus_watch_id;
/* Initialisation */
gst_init(&argc, &argv);
loop = g_main_loop_new(NULL, FALSE);
GError *error = NULL;
data.pipeline = gst_parse_launch("concat name=c ! filesink location=program.wav appsrc name=src_00 ! oggdemux ! opusdec ! audioconvert ! audioresample ! audio/x-raw,format=S16LE,channels=1,rate=16000 ! queue ! c.", &error);
if (!data.pipeline)
{
g_printerr("Pipeline could not be created. Exiting.\n");
return -1;
}
data.app_source = (G_TYPE_CHECK_INSTANCE_CAST((gst_bin_get_by_name(GST_BIN(data.pipeline), "src_00")), GST_TYPE_APP_SRC, GstAppSrc));
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);
/* we add a message handler */
bus = gst_pipeline_get_bus(GST_PIPELINE(data.pipeline));
bus_watch_id = gst_bus_add_watch(bus, bus_call, NULL);
gst_object_unref(bus);
file = fopen("junk.wav", "rb");
/* Set the pipeline to "playing" state*/
g_print("Now playing");
gst_element_set_state(data.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(data.pipeline, GST_STATE_NULL);
g_print("Deleting pipeline\n");
gst_object_unref(GST_OBJECT(data.pipeline));
g_source_remove(bus_watch_id);
g_main_loop_unref(loop);
return 0;
}
I would have expected that the demux failure would follow to the message bus, but it is not. How can I listen to such errors ?
I've tried with other pipelines that uses decodebin and I get the error messages on the message bus. The following pipeline works as expected:
gst_parse_launch("concat name=c ! filesink location=program.wav appsrc name=src_00 ! decodebin ! audioconvert ! audioresample ! audio/x-raw,format=S16LE,channels=1,rate=16000 ! queue ! c.", &error);
GStreamer version: 1.8.3
OS: Ubuntu 16.04
The issue seems to be resolved in Gstreamer 1.14. After updating I now get an error message on the message bus:
Message: Error: from oggdemux0 Could not demultiplex stream.
Error Code: GST_STREAM_ERROR_DEMUX

Issue with Gstreamer under Qt in c++ : flow error

I'm a beginner in Gstreamer, I need to get the frame from an UDP camera and convert it to an cv::Mat(OpenCV).
I run my camera stream like this :
gst-launch-1.0 v4l2src device=/dev/video0 ! \
h264parse ! rtph264pay ! udpsink host=XXX.XXX.XXX.XXX port=5000
And in another terminal I can get the stream like this (and it works):
gst-launch-1.0 udpsrc port=5000 caps='application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264' ! \
rtph264depay ! avdec_h264 ! autovideosink
So in my C++ code here is my C++ code :
GstFlowReturn
new_preroll(GstAppSink *appsink, gpointer data) {
g_print ("Got preroll!\n");
return GST_FLOW_OK;
}
GstFlowReturn
new_sample(GstAppSink *appsink, gpointer data) {
static int framecount = 0;
framecount++;
GstSample *sample = gst_app_sink_pull_sample(appsink);
GstCaps *caps = gst_sample_get_caps(sample);
GstBuffer *buffer = gst_sample_get_buffer(sample);
const GstStructure *info = gst_sample_get_info(sample);
// ---- Read frame and convert to opencv format ---------------
GstMapInfo map;
gst_buffer_map (buffer, &map, GST_MAP_READ);
// convert gstreamer data to OpenCV Mat, you could actually
// resolve height / width from caps...
Mat frame(Size(320, 240), CV_8UC3, (char*)map.data, Mat::AUTO_STEP);
int frameSize = map.size;
// TODO: synchronize this....
frameQueue.push_back(frame);
gst_buffer_unmap(buffer, &map);
// ------------------------------------------------------------
// print dot every 30 frames
if (framecount%30 == 0) {
g_print (".");
}
// show caps on first frame
if (framecount == 1) {
g_print ("%s\n", gst_caps_to_string(caps));
}
gst_sample_unref (sample);
return GST_FLOW_OK;
}
static gboolean
my_bus_callback (GstBus *bus, GstMessage *message, gpointer data) {
g_print ("Got %s message\n", GST_MESSAGE_TYPE_NAME (message));
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_ERROR: {
GError *err;
gchar *debug;
gst_message_parse_error (message, &err, &debug);
g_print ("Error: %s\n", err->message);
g_error_free (err);
g_free (debug);
break;
}
case GST_MESSAGE_EOS:
/* end-of-stream */
break;
default:
/* unhandled message */
break;
}
/* we want to be notified again the next time there is a message
* on the bus, so returning TRUE (FALSE means we want to stop watching
* for messages on the bus and our callback should not be called again)
*/
return TRUE;
}
int
main (int argc, char *argv[])
{
GError *error = NULL;
gst_init (&argc, &argv);
gchar *descr = g_strdup(
"udpsrc port=5000 ! "
"caps=application/x-rtp, media=(string)video, clock-rate=(int)9000, encoding-name=(string)H264 ! "
"rtph264depay ! "
"avdec_h264 ! "
"videoconvert ! "
"appsink name=sink "
);
GstElement *pipeline = gst_parse_launch (descr, &error);
if (pipeline== NULL) {
g_print ("could not construct pipeline: %s\n", error->message);
g_error_free (error);
exit (-1);
}
/* get sink */
GstElement *sink = gst_bin_get_by_name (GST_BIN (pipeline), "sink");
gst_app_sink_set_emit_signals((GstAppSink*)sink, true);
gst_app_sink_set_drop((GstAppSink*)sink, true);
gst_app_sink_set_max_buffers((GstAppSink*)sink, 1);
GstAppSinkCallbacks callbacks = { NULL, new_preroll, new_sample };
gst_app_sink_set_callbacks (GST_APP_SINK(sink), &callbacks, NULL, NULL);
GstBus *bus;
guint bus_watch_id;
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
bus_watch_id = gst_bus_add_watch (bus, my_bus_callback, NULL);
gst_object_unref (bus);
GstStateChangeReturn test=gst_element_set_state (GST_ELEMENT (pipeline), GST_STATE_PLAYING);
qDebug() <<test<< " this is the test";
namedWindow("edges",1);
while(1) {
g_main_iteration(false);
// TODO: synchronize...
if (frameQueue.size() >10) {
// this lags pretty badly even when grabbing frames from webcam
Mat frame = frameQueue.front();
imshow("edges", frame);
cv::waitKey(30);
frameQueue.clear();
}
}
gst_element_set_state (GST_ELEMENT (pipeline), GST_STATE_NULL);
gst_object_unref (GST_OBJECT (pipeline));
return 0;
}
And I get the ERROR :
Error: Internal data flow error.
I think it is from my declaration of my pipeline, but I can't find what's wrong with it.
Any suggestions?
You have a ! after udpsrc port=5000. That one is not present in your original pipeline. I haven't checked any further, maybe print out the pipeline and double check if its the desired one.

running the gstreamer pipeline (not able to get video and audio data in the callback)

I'm a newbie to gstreamer and I wanted to get the audio and video both buffers from a 3gp file and do some processing in the callback.
(I'm starting my pipeline into a separate thread, pipeline gives audio buffers in a callback AudioCallback and video buffers in VideoCallback.)
This is how my pipeline looks:
GstElement* audioQueue;//global variable , needed in on_pad_added (cant pass both while connecting demuxer to callback)
GstElement* videoQueue;//global variable , needed in on_pad_added (cant pass both while connecting demuxer to callback)
//static functions
static gboolean
bus_call (GstBus* bus, GstMessage* msg, gpointer data)
{
GMainLoop* loop = (GMainLoop*) data;
switch (GST_MESSAGE_TYPE (msg))
{
case GST_MESSAGE_EOS:
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;
}
static void link_two_elements(GstElement* src_element, GstElement* sink_element)
{
if(!gst_element_link(src_element, sink_element))
g_printerr ("Linking Error");
}
static void
on_pad_added (GstElement *element,
GstPad *pad,
gpointer data)
{
GstCaps *caps;
GstStructure *str;
gchar *tex;
GstPad* sinkpad;
/* check media type */
caps = gst_pad_get_caps (pad);
str = gst_caps_get_structure (caps, 0);
tex = (gchar*)gst_structure_get_name(str);
if(g_strrstr(tex,"audio"))
{
//GstElement *audioQueue = (GstElement *) data;
sinkpad = gst_element_get_static_pad (audioQueue, "sink");
if(sinkpad)
{
GstPadLinkReturn linkReturn = gst_pad_link (pad, sinkpad);
gst_object_unref (sinkpad);
}
}
if(g_strrstr(tex,"video"))
{
//GstElement *videoQueue = (GstElement *) data;
sinkpad = gst_element_get_static_pad (videoQueue, "sink");
GstPadLinkReturn linkReturn = gst_pad_link (pad, sinkpad);
gst_object_unref (sinkpad);
}
}
void runPipeline()
{
GMainLoop *loop;
GstElement *__pPipeline, *source, *demuxer, *audioDecoder, *audioConverter, *audioresample, /**audioQueue,*/ *audioSink, *videoDecoder, *videoSink, /**videoQueue,*/ *ffmpegcolorspace, *videoscale;
GstBus* bus;
//Initialisation
gst_init (null,null);
loop = g_main_loop_new (NULL, FALSE);
// Create gstreamer elements
__pPipeline = gst_pipeline_new("test_appsink");
source = gst_element_factory_make ("filesrc", "file-source");
demuxer = gst_element_factory_make("qtdemux", "demuxer");
//audioDecoder = gst_element_factory_make("ffdec_mp3", "audioDecoder");
audioDecoder = gst_element_factory_make("decodebin", "audioDecoder");
audioConverter = gst_element_factory_make("audioconvert", "audioConverter");
audioresample = gst_element_factory_make("audioresample", "audioresample");
audioSink = gst_element_factory_make("appsink", "audioSink");
audioQueue = gst_element_factory_make("queue2", "audioQueue");
//videoDecoder = gst_element_factory_make("ffdec_h264", "videoDecoder");
videoQueue = gst_element_factory_make("queue2", "videoQueue");
videoDecoder = gst_element_factory_make("decodebin ", "videoDecoder");
ffmpegcolorspace = gst_element_factory_make("ffmpegcolorspace", "ffmpegcolorspace");
videoscale = gst_element_factory_make("videoscale", "videoscale");
videoSink = gst_element_factory_make("appsink", "videoSink");
//appsink = gst_element_factory_make("appsink", "sink-buffer");
if (!__pPipeline || !source || !demuxer || !audioDecoder || !audioConverter ||!audioresample || !audioSink || !videoSink || !audioQueue || !videoQueue || !videoDecoder || !ffmpegcolorspace || !videoscale )
{
//return -1;
}
//we set the input filename to the source element
g_object_set (G_OBJECT (source), "location", "/etc/20000101-161404.3gp", NULL);
//Make appsink emit the "new-preroll" and "new-buffer" signals.
gst_app_sink_set_emit_signals ((GstAppSink*) audioSink, TRUE);
gst_app_sink_set_emit_signals ((GstAppSink*) videoSink, TRUE);
//we add a message handler
bus = gst_pipeline_get_bus (GST_PIPELINE (__pPipeline));
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 (__pPipeline),
source, demuxer, videoDecoder, audioDecoder, audioConverter, audioresample, audioSink, videoSink,
audioQueue, videoQueue, ffmpegcolorspace, videoscale, NULL);
//link source and demuxer seperately
link_two_elements(source, demuxer);
//link rest of the elements
int retValVideoLinking = (int)gst_element_link_many (videoQueue, videoDecoder, ffmpegcolorspace, videoscale, videoSink, NULL);
int retValAudioLinking = (int)gst_element_link_many (audioQueue, audioDecoder, audioConverter, audioresample, audioSink, NULL);
gulong sigConRet = g_signal_connect (demuxer, "pad-added", G_CALLBACK (on_pad_added), null);
_ArAudioIn audioInstance = _ArAudioIn::GetArAudioInstance();
g_signal_connect (videoSink, "new-buffer", G_CALLBACK (AudioCallback), null);//AudioCallback static API
g_signal_connect (audioSink, "new-buffer", G_CALLBACK (VideoCallback), null);//VideoCallback static API
//Set the pipeline to "playing" state
GstStateChangeReturn state = gst_element_set_state (__pPipeline, GST_STATE_PLAYING);
g_main_loop_run (loop);
return null;
}
I'm just getting a single video buffer in my Videocallback and also in the on_pad_addded : I'm getting a linking err for audio pad linking.
GST_PAD_LINK_NOFORMAT = -4,
I'm trying to link the queue's sink pad to the pad recieved in on_pad_added, same is working for video but not for audio.
If anybody has any idea about this then please give me some pointers to get rid off this err and make this pipeline work.
It would be nice if you cleanup you code before asking us to debug it. As a general advice, check the return values and either log a warning or simply exit(1) to ensure that your pipeline setup works (E.g. in the pad_added handler). I'd also start using a normal video and audiosink to check that it plays.
Finally, it is usually a bad idea to pull out data from the pipleine. Perhaps you could tell what you want to do with the data once you have it in your callback, so that we can give better advice.