Gstreamer debug pipeline c++ - c++

I am trying to transcode a gstreamer bash script to c++ code, y but I am not able to save de debuggin log into a file
This is my code
int main(int argc, char *argv[])
{
YoctoLinuxSystem* hola;
//hola->YoctoLinuxSystem();
CustomData DataTest, DataDvi;
GstBus *bus;
GstMessage *msg;
GstStateChangeReturn ret;
GMainLoop *loop;
//vector<string> lines = YoctoLinuxSystem::getCmdOutputAsLines("./scripts/get_system_temps.sh");
gst_init (&argc, &argv);
loop = g_main_loop_new (NULL, FALSE);
/******************************/
/****AJUSTES GSTREAMER TEST****/
/******************************/
DataTest.source = gst_element_factory_make ("videotestsrc", "source");
DataTest.capsfilter = gst_element_factory_make ("capsfilter","caps");
DataTest.sink = gst_element_factory_make ("imxipuvideosink", "sink");
DataTest.pipeline = gst_pipeline_new ("test-pipeline");
if (!DataTest.pipeline || !DataTest.source || !DataTest.capsfilter || !DataTest.sink) {
g_printerr ("Not all elements could be created.\n");
return -1;
}
/* Modify the source's properties */
g_object_set (DataTest.source, "pattern", 0, NULL);
g_object_set(DataTest.capsfilter, "caps", gst_caps_new_simple("video/x-raw", "framerate", GST_TYPE_FRACTION, 25, 1,"width", G_TYPE_INT, 1920, "height", G_TYPE_INT, 1080, "format", G_TYPE_STRING, "RGB", NULL), NULL);
/* Build the pipeline */
gst_bin_add_many (GST_BIN (DataTest.pipeline), DataTest.source, DataTest.capsfilter, DataTest.sink, NULL);
if (gst_element_link (DataTest.source, DataTest.capsfilter) != TRUE) {
g_printerr ("Elements source-caps could not be linked.\n");
gst_object_unref (DataTest.pipeline);
return -1;
}
if (gst_element_link (DataTest.capsfilter, DataTest.sink) != TRUE) {
g_printerr ("Elements caps-sink could not be linked.\n");
gst_object_unref (DataTest.pipeline);
return -1;
}
gst_element_link_many (DataTest.source, DataTest.capsfilter, DataTest.sink, NULL);
/******************************/
/****AJUSTES GSTREAMER DVI****/
/******************************/
DataDvi.source = gst_element_factory_make ("v4l2src", "source");
DataDvi.sink = gst_element_factory_make ("imxipuvideosink", "sink");
DataDvi.pipeline = gst_pipeline_new ("test-pipeline");
if (!DataDvi.pipeline || !DataDvi.source || !DataDvi.sink) {
g_printerr ("Not all elements could be created.\n");
return -1;
}
/* Modify the source's properties */
g_object_set (DataDvi.source, "device", "/dev/video0", NULL);
/* Build the pipeline */
gst_bin_add_many (GST_BIN (DataDvi.pipeline), DataDvi.source, DataDvi.sink, NULL);
if (gst_element_link (DataDvi.source, DataDvi.sink) != TRUE) {
g_printerr ("Elements caps-sink could not be linked.\n");
gst_object_unref (DataDvi.pipeline);
return -1;
}
gst_element_link_many (DataDvi.source, DataDvi.sink, NULL);
GST_DEBUG=2;
ifstream fileread;
// fileread.open("/var/log/data.log");
while(1)
{
ifstream fileread("/var/log/data.log");
if (!fileread.good())
{
/* Start playing */
//g_print ("Now playing: \n");
gst_element_set_state (DataDvi.pipeline, GST_STATE_PAUSED);
ret = gst_element_set_state (DataTest.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 (DataTest.pipeline);
return -1;
}
}
else
{
gst_element_set_state (DataTest.pipeline, GST_STATE_PAUSED);
ret = gst_element_set_state (DataDvi.pipeline, GST_STATE_PLAYING);
/*HERE I NEED TO KNOW THE DEBUG OF THE PIPELINE*/
if (ret == GST_STATE_CHANGE_FAILURE) {
g_printerr ("Unable to set the pipeline to the playing state.\n");
gst_object_unref (DataDvi.pipeline);
return -1;
}
}
}
g_print ("SALE\n");
return 0;
}
I am using gstreamer-1.0 library and I have seen that I need use GST_DEBUG_FILE but I do not know how call these functions from c++.
Thanks for the help!

There is a rich API behind the debugging and logging infrastructure. Take a look at this documentation.
For example, you might set some default values when the environment variables have not been setup.
if (!gst_debug_is_active()) {
gst_debug_set_active(TRUE);
GstDebugLevel dbglevel = gst_debug_get_default_threshold();
if (dbglevel < GST_LEVEL_ERROR) {
dbglevel = GST_LEVEL_ERROR;
gst_debug_set_default_threshold(dbglevel);
}
}
For that, the previous poster correctly advised you about the environment variable, GST_DEBUG_FILE. You could do putenv() before gst_init().
If you want to get fancy, you could can replace the default log function, gst_debug_log_default(). You do this by adding your own via, gst_debug_add_log_function(); then remove the default, gst_debug_remove_log_function(gst_debug_log_default). If you simply wish to change the file, then gst_debug_add_log_function (gst_debug_log_default, log_file, NULL), for some open and ready FILE* log_file.

GST_DEBUG_FILE is environment variable, so it has nothing to do with C++.
You could just use something like
export GST_DEBUG_FILE=~/gst.log
before run your application. Or add something like this to your bash startup script.

Related

gstreamer pipeline from cam to file C code ends up with empty output file

I have an USB camera. I have working terminal commands to record or display fullHD video and to save one 4k image. I would like to handle it all via C++ app. If we will concentrate on the video-saving:
gst-launch-1.0 v4l2src device=/dev/video0 num-buffers=900! image/jpeg, width=1920, height=1080, io-mode=4 ! imxvpudec ! imxvpuenc_mjpeg ! avimux ! filesink location=/mnt/ssd/test.avi
will save 900frames (aka 30s) of video. I would like to have C++ code to record indefinetly (in future maybe in hour-long segments) until I (the app) tell it to end.
I came up with
struct {
GstElement *pipeline_sink, *source, *appsink;
GstElement *pipeline_src, *appsrc, *decoder, *mux, *sink, *encoder;
} usbCam::mGstData;
int usbCam::gstInit(){
GstCaps *caps;
GstStateChangeReturn ret;
// Initialize GStreamer
if (!gst_is_initialized()) {
setenv("GST_DEBUG", ("*:" + std::to_string(3)).c_str(), 1);
gst_init(nullptr, nullptr);
}
// Create the elements
mGstData.source = gst_element_factory_make ("v4l2src", "source");
g_object_set (mGstData.source, "device", "/dev/video0", NULL);
mGstData.pipeline_sink = gst_pipeline_new ("pipeline_sink");
caps = gst_caps_new_any();
gst_app_sink_set_caps(GST_APP_SINK(mGstData.appsink), caps);
gst_caps_unref (caps);
gst_app_sink_set_emit_signals(GST_APP_SINK(mGstData.appsink), true);
// Build the pipeline
gst_bin_add_many (GST_BIN (mGstData.pipeline_sink), mGstData.source, mGstData.appsink, NULL);
if (gst_element_link_many(mGstData.source, mGstData.appsink, NULL) != TRUE) {
g_printerr ("Elements could not be linked.\n");
gst_object_unref (mGstData.pipeline_sink);
return -1;
}
return 0;
}
int usbCam::videoStart(){
GstCaps *caps;
GstStateChangeReturn ret;
if (!mGstData.pipeline_sink || !mGstData.source) {
g_printerr ("Not all elements could be created.\n");
return -1;
}
mGstData.appsrc = gst_element_factory_make ("appsrc", "appsrc");
mGstData.decoder = gst_element_factory_make ("imxvpudec", "transform_enc");
mGstData.mux = gst_element_factory_make ("avimux", "avimux");
mGstData.sink = gst_element_factory_make ("filesink", "sink");
g_object_set (mGstData.sink, "location", "/mnt/ssd/videoTest.avi", NULL);
mGstData.pipeline_src = gst_pipeline_new ("pipeline_src");
if (!mGstData.pipeline_src || !mGstData.appsrc || !mGstData.decoder || !mGstData.mux || !mGstData.sink) {
g_printerr ("Not all elements could be created.\n");
return -1;
}
caps = gst_caps_new_simple ("image/jpeg",
"width", G_TYPE_INT, 1920,
"height", G_TYPE_INT, 1080,
"io-mode", G_TYPE_INT, 4,
NULL);
gst_app_src_set_caps(GST_APP_SRC(mGstData.appsrc), caps);
gst_caps_unref (caps);
gst_app_src_set_duration(GST_APP_SRC(mGstData.appsrc), GST_TIME_AS_MSECONDS(80));
gst_app_src_set_stream_type(GST_APP_SRC(mGstData.appsrc), GST_APP_STREAM_TYPE_STREAM);
gst_app_src_set_latency(GST_APP_SRC(mGstData.appsrc), -1, 0);
gst_bin_add_many (GST_BIN (mGstData.pipeline_src), mGstData.appsrc, mGstData.decoder, mGstData.sink, NULL);
if (gst_element_link_many(mGstData.appsrc, mGstData.decoder, mGstData.sink, NULL) != TRUE) {
g_printerr ("Elements could not be linked.\n");
gst_object_unref (mGstData.pipeline_src);
return -1;
}
ret = gst_element_set_state (mGstData.pipeline_src, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE) {
g_printerr ("Unable to set the pipeline to the playing state.\n");
gst_object_unref (mGstData.pipeline_src);
return -1;
}
return 0;
}
int usbCam::videoEnd(){
{
gst_app_src_end_of_stream(GST_APP_SRC(mGstData.appsrc));
usleep(500000);
gst_element_set_state (mGstData.pipeline_src, GST_STATE_NULL);
gst_object_unref (mGstData.pipeline_src);
return 0;
}
Now, this code runs. No error in the output, one warning though:
(GLib-GObject-WARNING **: 17:51:34.132: g_object_set_is_valid_property: object class 'GstSplitMuxSink' has no property named 'h}\x9fe h\xe6a_no_\xc1')
.
What actually bothers me is the output file. It is created, but it is an empty file with 0b size. Can anyone point me in the direction of the proper fix?
Edit: Today I came up with two other attempts. The firs one is not that different from the one already posted here. The second gives me pipeline with wrong parameters (different FPS) and I am unable to correctly stop it so that the file have correct EOF.
GstElement *pipeline;
GstBus *bus;
GstMessage *msg;
std::string command = "v4l2src device=/dev/video0 ! image/jpeg, width=1920, height=1080, io-mode=4 ! imxvpudec ! imxvpuenc_mjpeg ! avimux ! filesink location = /mnt/ssd/testPipeline.avi";
/* Build the pipeline */
pipeline =
gst_parse_launch
(command.c_str(),
NULL);
/* Start playing */
gst_element_set_state (pipeline, GST_STATE_PLAYING);
/* Wait until error or EOS */
bus = gst_element_get_bus (pipeline);
msg =
gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE, GstMessageType(
GST_MESSAGE_ERROR | GST_MESSAGE_EOS));
/* Free resources */
if (msg != NULL)
gst_message_unref (msg);
gst_object_unref (bus);
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (pipeline);
EDIT2:
OK now my code looks like this:
GstElement *pipeline;
GstElement *tee; //in the future I would like to save video and images AND stream or use thi pipeline data internally.
void gstFail(const gchar* message){
g_printerr(message);
gst_object_unref (pipeline);
return;
}
void videoStart(std::string path){
if (!gst_is_initialized()) {
setenv("GST_DEBUG", ("*:" + std::to_string(3)).c_str(), 1);
gst_init(nullptr, nullptr);
}
GstCaps *caps;
GstStateChangeReturn ret;
GstElement *source, *muxer, *sink;
source = gst_element_factory_make ("v4l2src", "source");
g_object_set (source, "device", mVideoDevice.toStdString().c_str(), NULL);
muxer = gst_element_factory_make ("avimux", "avimux");
tee = gst_element_factory_make("tee", "tee");
sink = gst_element_factory_make ("filesink", "sink");
g_object_set (sink, "location", path.c_str(), NULL);
pipeline = gst_pipeline_new ("pipeline_src");
if (!pipeline || !source || !muxer || !sink) {
g_printerr ("Not all elements could be created.\n");
return;
}
caps = gst_caps_new_simple ("image/jpeg",
"width", G_TYPE_INT, 1920,
"height", G_TYPE_INT, 1080,
"io-mode", G_TYPE_INT, 4,
"framerate", GST_TYPE_FRACTION, 30, 1,
"pixel-aspect-ratio", GST_TYPE_FRACTION, 1,1,
"interlace-mode", G_TYPE_STRING, "progresive",
NULL);
gst_bin_add_many (GST_BIN (pipeline), source, muxer,tee, sink, NULL);
if (gst_element_link_filtered(source, muxer, caps) != TRUE) {
gst_caps_unref (caps);
gstFail("Elements could not be linked or caps set.\n");
return;
}
gst_caps_unref (caps);
if (gst_element_link_many(muxer,tee, sink, NULL) != TRUE) {
gstFail("Elements could not be linked or caps set.\n");
return;
}
ret = gst_element_set_state (pipeline, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE) {
gstFail("Unable to set the pipeline to the playing state.\n");
return;
}
return;
}
void videoEnd(void)
{
GstMessage *message = gst_message_new_eos(&pipeline->object);
gst_bus_post(pipeline->bus, message);
/* Free resources */
if (message != NULL)
gst_message_unref (message);
gst_element_change_state(pipeline, GST_STATE_CHANGE_PLAYING_TO_PAUSED);
gst_element_change_state(pipeline, GST_STATE_CHANGE_PAUSED_TO_READY);
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref(pipeline);
}
void takeImage(std::string path){
GstElement *sink = gst_element_factory_make("multifilesink", "multifilesink");
g_object_set (sink, "location", path.c_str(), NULL);
gst_bin_add_many (GST_BIN (pipeline), sink, NULL);
if (gst_element_link_many(tee, sink, NULL) != TRUE) {
gstFail("Elements could not be linked or caps set.\n");
return;
}
return;
}
This saves the video ALMOST ok (VLC does not display correct lenght. But when I see the video file properties via Nautilus in Ubuntu the correct lenght is displayed and the video is playable). It does not save the pictures.
OK, so here's how I solved it: my initial pipeline is split with tee element into two sinks: the original sink that saves the video and appsink. In the callback functuion for the appsink I create new pipeline and push the frame any time I want to save the image. Basically:
...
int saveSampleFromAppsinkJpeg( GstSample *sample){
if (!shouldSaveImage) {
return -2;
}
if (capturing){
return -3;
}
std::thread([=]{
capturing = true;
GstStateChangeReturn ret;
GstElement *appsrc = gst_element_factory_make ("appsrc", "appsrc");
GstElement *sink = gst_element_factory_make ("multifilesink", "sink");
g_object_set (sink, "location", "some/path", NULL);
GstElement *pipeline_img = gst_pipeline_new ("pipeline_img");
if (!pipeline_img || !appsrc || !sink) {
g_printerr ("Not all elements could be created.\n");
capturing = false;
return -1;
}
gst_app_src_set_caps(GST_APP_SRC(appsrc), caps);
gst_app_src_set_duration(GST_APP_SRC(appsrc), GST_TIME_AS_MSECONDS(80)); // TODO 80
gst_app_src_set_stream_type(GST_APP_SRC(appsrc), GST_APP_STREAM_TYPE_STREAM);
gst_app_src_set_latency(GST_APP_SRC(appsrc), -1, 0);
gst_bin_add_many (GST_BIN (pipeline_img), appsrc, sink, NULL);
if (gst_element_link_many(appsrc, sink, NULL) != TRUE) {
g_printerr ("Elements could not be linked.\n");
gst_object_unref (pipeline_img);
capturing = false;
return -1;
}
ret = gst_element_set_state (pipeline_img, 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_img);
capturing = false;
return -1;
}
//push the image in the pipeline
GstFlowReturn status = GstFlowReturn::GST_FLOW_OK;
status = gst_app_src_push_sample(GST_APP_SRC(appsrc), sample);
if (status != GstFlowReturn::GST_FLOW_OK) g_printerr ("Sample for saving image not pushed.\n");
status = gst_app_src_end_of_stream(GST_APP_SRC(appsrc));
if (status != GstFlowReturn::GST_FLOW_OK) g_printerr ("EOS for saving image not pushed.\n");
//end the pipeline
usleep(500000); // Important
GstMessage *message = gst_message_new_eos(&pipeline_img->object);
gst_bus_post(pipeline_img->bus, message);
/* Free resources */
if (message != NULL)
gst_message_unref (message);
gst_element_set_state (pipeline_img, GST_STATE_PAUSED);
gst_element_set_state (pipeline_img, GST_STATE_NULL);
gst_object_unref (pipeline_img);
shouldSaveImage = false;
capturing = false;
return 1;
}).detach();
return 1;
}
static GstFlowReturn new_sample_jpeg(GstElement * elt)
{
GstSample *sample;
GstBuffer *buffer;
GstMemory *memory;
GstFlowReturn ret = GST_FLOW_OK;
// get the sample from appsink
sample = gst_app_sink_pull_sample (GST_APP_SINK (elt));
buffer = gst_sample_get_buffer (sample);
if (buffer != NULL) {
memory = gst_buffer_get_memory (buffer, 0);
if (memory != NULL) {
//now all data are image data. If image wanted->image save!
if (wantToSave) saveSampleFromAppsinkJpeg(sample);
}
...
}
}
void startVideo(){
if (!gst_is_initialized()) {
setenv("GST_DEBUG", ("*:" + std::to_string(3)).c_str(), 1);
gst_init(nullptr, nullptr);
}
GstStateChangeReturn ret;
GstElement *source, *muxer, *sink, *queue_rcr, *queue_app, *appsink;
source = gst_element_factory_make ("v4l2src", "source");
g_object_set (source, "device", "/dev/video1", NULL);
muxer = gst_element_factory_make ("avimux", "avimux");
tee = gst_element_factory_make("tee", "tee");
sink = gst_element_factory_make ("filesink", "sink");
queue_rcr = gst_element_factory_make ("queue", "record_queue");
queue_app = gst_element_factory_make ("queue", "app_queue");
appsink = gst_element_factory_make("appsink", "appsink");
g_object_set (sink, "location", path.toStdString().c_str(), NULL);
pipeline = gst_pipeline_new ("pipeline_src");
if (!pipeline || !source || !muxer || !sink || !queue_rcr || !appsink) {
g_printerr ("Not all elements could be created.\n");
return;
}
caps = gst_caps_new_simple ("image/jpeg",
"width", G_TYPE_INT, 1920,
"height", G_TYPE_INT, 1080,
"io-mode", G_TYPE_INT, 4,
"framerate", GST_TYPE_FRACTION, 30, 1,
"pixel-aspect-ratio", GST_TYPE_FRACTION, 1,1,
"interlace-mode", G_TYPE_STRING, "progresive",
NULL);
gst_bin_add_many (GST_BIN (pipeline), source, muxer,tee, sink,queue_rcr, appsink, queue_app, NULL);
if (gst_element_link_filtered(source, tee, caps) != TRUE) {
//failhandling
}
if (gst_element_link_many(tee, queue_rcr, muxer, sink, NULL) != TRUE) {
//failhandling
}
if (gst_element_link_many(tee, queue_app, appsink, NULL) != TRUE) {
//failhandling
}
gst_app_sink_set_emit_signals(GST_APP_SINK(appsink), true);
g_signal_connect (appsink, "new-sample", G_CALLBACK (new_sample_jpeg));
ret = gst_element_set_state (pipeline, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE) {
//failhandling
}
// Start playing
recording = true;
return;
}

Gstreamer recording video with audio

I'm trying to record on a file a video from my webcam along with audio using Gstreamer on my Ubuntu 16 machine through glib library.
I'm able to watch the video streaming from the webcam through these code lines
#include <gst/gst.h>
int main(int argc, char *argv[]) {
GstElement *pipeline, *source, *sink, *convert;
GstBus *bus;
GstMessage *msg;
GstStateChangeReturn ret;
/* Initialize GStreamer */
gst_init (&argc, &argv);
/* Create the elements */
source = gst_element_factory_make ("v4l2src", "source");
sink = gst_element_factory_make ("autovideosink", "sink");
convert =gst_element_factory_make("videoconvert","convert");
//convert = gst_element_factory_make ("audioconvert", "convert");
//sink = gst_element_factory_make ("autoaudiosink", "sink");
/* Create the empty pipeline */
pipeline = gst_pipeline_new ("test-pipeline");
if (!pipeline || !source || !sink || !convert) {
g_printerr ("Not all elements could be created.\n");
return -1;
}
/*set der source*/
g_object_set (source, "device", "/dev/video0", NULL);
/* Build the pipeline */
gst_bin_add_many (GST_BIN (pipeline), source, sink, convert, NULL);
if (gst_element_link (convert, sink) != TRUE) {
g_printerr ("Elements could not be linked confert sink.\n");
gst_object_unref (pipeline);
return -1;
}
if (gst_element_link (source, convert) != TRUE) {
g_printerr ("Elements could not be linked source -convert.\n");
gst_object_unref (pipeline);
return -1;
}
/* 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,(GstMessageType) (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;
}
and to capture audio from microphone and listen it through the speakers using these code lines
#include <gst/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;
}
/* Main function for audio pipeline initialization and looping streaming process */
gint
main (gint argc, gchar **argv) {
GMainLoop *loop;
GstElement *pipeline, *audio_source, *sink;
GstBus *bus;
guint bus_watch_id;
GstCaps *caps;
gboolean ret;
/* Initialization of gstreamer */
gst_init (&argc, &argv);
loop = g_main_loop_new (NULL, FALSE);
/* Elements creation */
pipeline = gst_pipeline_new ("audio_stream");
audio_source = gst_element_factory_make ("alsasrc", "audio_source");
sink = gst_element_factory_make ("alsasink", "audio_sink");
// video_source = gst_element_factory_make ("v4l2src", "source");
// video_sink = gst_element_factory_make ("autovideosink", "sink");
// video_convert= gst_element_factory_make("videoconvert","convert");
if (!pipeline) {
g_printerr ("Audio: Pipeline couldn't be created\n");
return -1;
}
if (!audio_source) {
g_printerr ("Audio: alsasrc couldn't be created\n");
return -1;
}
if (!sink) {
g_printerr ("Audio: Output file couldn't be created\n");
return -1;
}
g_object_set (G_OBJECT (audio_source), "device", "hw:1,0", NULL);
g_object_set (G_OBJECT (sink), "device", "hw:1,0", NULL);
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
bus_watch_id = gst_bus_add_watch (bus, bus_call, loop);
gst_object_unref (bus);
gst_bin_add_many (GST_BIN(pipeline), audio_source, sink, NULL);
caps = gst_caps_new_simple ("audio/x-raw", "format", G_TYPE_STRING, "S16LE", "layout", G_TYPE_STRING, "interleaved", "rate", G_TYPE_INT, (int)44100, "channels", G_TYPE_INT, (int)2, NULL);
ret = gst_element_link_filtered (audio_source, sink, caps);
if (!ret) {
g_print ("audio_source and sink couldn't be linked\n");
gst_caps_unref (caps);
return FALSE;
}
gst_element_set_state (pipeline, GST_STATE_PLAYING);
g_print ("streaming...\n");
g_main_loop_run (loop);
g_print ("Returned, stopping stream\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;
}
What i really don't understand is how to get video from the webcam and audio from my alsa hw at the same time and save them into a file (such as .mp4 for ex). Can anyone help me? I tried to find something useful, but there's nothing on the board. In addition, it would be really appreciate also how to save just the video stream or just the audio stream in separated files.
UPDATE
I looked again to the tutorials and to the git link gave by #nayana, so i tried myself to code something. I have two results:
#include <string.h>
#include <gst/gst.h>
#include <signal.h>
#include <unistd.h>
#include <stdlib.h>
static GMainLoop *loop;
static GstElement *pipeline;
static GstElement *muxer, *sink;
static GstElement *src_video, *encoder_video, *queue_video;
static GstElement *src_audio, *encoder_audio, *queue_audio;
static GstBus *bus;
static gboolean
message_cb (GstBus * bus, GstMessage * message, gpointer user_data)
{
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_ERROR:{
GError *err = NULL;
gchar *name, *debug = NULL;
name = gst_object_get_path_string (message->src);
gst_message_parse_error (message, &err, &debug);
g_printerr ("ERROR: from element %s: %s\n", name, err->message);
if (debug != NULL)
g_printerr ("Additional debug info:\n%s\n", debug);
g_error_free (err);
g_free (debug);
g_free (name);
g_main_loop_quit (loop);
break;
}
case GST_MESSAGE_WARNING:{
GError *err = NULL;
gchar *name, *debug = NULL;
name = gst_object_get_path_string (message->src);
gst_message_parse_warning (message, &err, &debug);
g_printerr ("ERROR: from element %s: %s\n", name, err->message);
if (debug != NULL)
g_printerr ("Additional debug info:\n%s\n", debug);
g_error_free (err);
g_free (debug);
g_free (name);
break;
}
case GST_MESSAGE_EOS:{
g_print ("Got EOS\n");
g_main_loop_quit (loop);
gst_element_set_state (pipeline, GST_STATE_NULL);
g_main_loop_unref (loop);
gst_object_unref (pipeline);
exit(0);
break;
}
default:
break;
}
return TRUE;
}
void sigintHandler(int unused) {
g_print("You ctrl-c-ed! Sending EoS");
gst_element_send_event(pipeline, gst_event_new_eos());
}
int main(int argc, char *argv[])
{
signal(SIGINT, sigintHandler);
gst_init (&argc, &argv);
pipeline = gst_pipeline_new(NULL);
src_video = gst_element_factory_make("v4l2src", NULL);
encoder_video = gst_element_factory_make("x264enc", NULL);
queue_video = gst_element_factory_make("queue", NULL);
src_audio = gst_element_factory_make ("alsasrc", NULL);
encoder_audio = gst_element_factory_make("lamemp3enc", NULL);
queue_audio = gst_element_factory_make("queue", NULL);
muxer = gst_element_factory_make("mp4mux", NULL);
sink = gst_element_factory_make("filesink", NULL);
if (!pipeline || !src_video || !encoder_video || !src_audio || !encoder_audio
|| !queue_video || !queue_audio || !muxer || !sink) {
g_error("Failed to create elements");
return -1;
}
g_object_set(src_audio, "device", "hw:1,0", NULL);
g_object_set(sink, "location", "video_audio_test.mp4", NULL);
gst_bin_add_many(GST_BIN(pipeline), src_video, encoder_video, queue_video,
src_audio, encoder_audio, queue_audio, muxer, sink, NULL);
gst_element_link_many (src_video,encoder_video,queue_video, muxer,NULL);
gst_element_link_many (src_audio,encoder_audio,queue_audio, muxer,NULL);
if (!gst_element_link_many(muxer, sink, NULL)){
g_error("Failed to link elements");
return -2;
}
loop = g_main_loop_new(NULL, FALSE);
bus = gst_pipeline_get_bus(GST_PIPELINE (pipeline));
gst_bus_add_signal_watch(bus);
g_signal_connect(G_OBJECT(bus), "message", G_CALLBACK(message_cb), NULL);
gst_object_unref(GST_OBJECT(bus));
gst_element_set_state(pipeline, GST_STATE_PLAYING);
g_print("Starting loop");
g_main_loop_run(loop);
return 0;
}
With this upon i am able to record the video from the cam, but the audio is recorded for just one second somewhere randomly during the recording and it gives me this error
ERROR: from element /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: Can't record audio fast enough
Additional debug info:
gstaudiobasesrc.c(869): gst_audio_base_src_create (): /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0:
Dropped 206388 samples. This is most likely because downstream can't keep up and is consuming samples too slowly.<br>
So i tried to add some setting and queues
#include <string.h>
#include <gst/gst.h>
#include <signal.h>
#include <unistd.h>
#include <stdlib.h>
static GMainLoop *loop;
static GstElement *pipeline;
static GstElement *muxer, *sink;
static GstElement *src_video, *encoder_video, *queue_video, *rate_video, *scale_video, *capsfilter_video;
static GstElement *src_audio, *encoder_audio, *queue_audio, *queue_audio2, *capsfilter_audio, *rate_audio;
static GstBus *bus;
static GstCaps *caps;
static gboolean
message_cb (GstBus * bus, GstMessage * message, gpointer user_data)
{
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_ERROR:{
GError *err = NULL;
gchar *name, *debug = NULL;
name = gst_object_get_path_string (message->src);
gst_message_parse_error (message, &err, &debug);
g_printerr ("ERROR: from element %s: %s\n", name, err->message);
if (debug != NULL)
g_printerr ("Additional debug info:\n%s\n", debug);
g_error_free (err);
g_free (debug);
g_free (name);
g_main_loop_quit (loop);
break;
}
case GST_MESSAGE_WARNING:{
GError *err = NULL;
gchar *name, *debug = NULL;
name = gst_object_get_path_string (message->src);
gst_message_parse_warning (message, &err, &debug);
g_printerr ("ERROR: from element %s: %s\n", name, err->message);
if (debug != NULL)
g_printerr ("Additional debug info:\n%s\n", debug);
g_error_free (err);
g_free (debug);
g_free (name);
break;
}
case GST_MESSAGE_EOS:{
g_print ("Got EOS\n");
g_main_loop_quit (loop);
gst_element_set_state (pipeline, GST_STATE_NULL);
g_main_loop_unref (loop);
gst_object_unref (pipeline);
exit(0);
break;
}
default:
break;
}
return TRUE;
}
void sigintHandler(int unused) {
g_print("You ctrl-c-ed! Sending EoS");
gst_element_send_event(pipeline, gst_event_new_eos());
}
int main(int argc, char *argv[])
{
signal(SIGINT, sigintHandler);
gst_init (&argc, &argv);
pipeline = gst_pipeline_new(NULL);
src_video = gst_element_factory_make("v4l2src", NULL);
rate_video = gst_element_factory_make ("videorate", NULL);
scale_video = gst_element_factory_make ("videoscale", NULL);
capsfilter_video = gst_element_factory_make ("capsfilter", NULL);
queue_video = gst_element_factory_make("queue", NULL);
encoder_video = gst_element_factory_make("x264enc", NULL);
src_audio = gst_element_factory_make ("alsasrc", NULL);
capsfilter_audio = gst_element_factory_make ("capsfilter", NULL);
queue_audio = gst_element_factory_make("queue", NULL);
rate_audio = gst_element_factory_make ("audiorate", NULL);
queue_audio2 = gst_element_factory_make("queue", NULL);
encoder_audio = gst_element_factory_make("lamemp3enc", NULL);
muxer = gst_element_factory_make("mp4mux", NULL);
sink = gst_element_factory_make("filesink", NULL);
if (!pipeline || !src_video || !rate_video || !scale_video || !capsfilter_video
|| !queue_video || !encoder_video || !src_audio || !capsfilter_audio
|| !queue_audio || !rate_audio || !queue_audio2 || !encoder_audio
|| !muxer || !sink) {
g_error("Failed to create elements");
return -1;
}
// Set up the pipeline
g_object_set(src_video, "device", "/dev/video0", NULL);
g_object_set(src_audio, "device", "hw:1,0", NULL);
g_object_set(sink, "location", "video_audio_test.mp4", NULL);
// video settings
caps = gst_caps_from_string("video/x-raw,format=(string)I420,width=480,height=384,framerate=(fraction)25/1");
g_object_set (G_OBJECT (capsfilter_video), "caps", caps, NULL);
gst_caps_unref (caps);
// audio settings
caps = gst_caps_from_string("audio/x-raw,rate=44100,channels=1");
g_object_set (G_OBJECT (capsfilter_audio), "caps", caps, NULL);
gst_caps_unref (caps);
// add all elements into the pipeline
gst_bin_add_many(GST_BIN(pipeline), src_video, rate_video, scale_video, capsfilter_video,
queue_video, encoder_video, src_audio, capsfilter_audio, queue_audio, rate_audio,
queue_audio2, encoder_audio, muxer, sink, NULL);
if (!gst_element_link_many (src_video,rate_video,scale_video, capsfilter_video,
queue_video, encoder_video, muxer,NULL))
{
g_error("Failed to link video elements");
return -2;
}
if (!gst_element_link_many (src_audio, capsfilter_audio, queue_audio, rate_audio,
queue_audio2, encoder_audio, muxer,NULL))
{
g_error("Failed to link audio elements");
return -2;
}
if (!gst_element_link_many(muxer, sink, NULL))
{
g_error("Failed to link elements");
return -2;
}
loop = g_main_loop_new(NULL, FALSE);
bus = gst_pipeline_get_bus(GST_PIPELINE (pipeline));
gst_bus_add_signal_watch(bus);
g_signal_connect(G_OBJECT(bus), "message", G_CALLBACK(message_cb), NULL);
gst_object_unref(GST_OBJECT(bus));
gst_element_set_state(pipeline, GST_STATE_PLAYING);
g_print("Starting loop");
g_main_loop_run(loop);
return 0;
}
This time the code doesnt record anything and give me the following error
ERROR: from element /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: Internal data flow error.
Additional debug info:
gstbasesrc.c(2948): gst_base_src_loop (): /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0:
streaming task paused, reason not-negotiated (-4)
Can you address me to fix the error?
Thanks in advance
What you need is the multiplexer - such GStreamer element that can merge two streams into one.
mp4, mkv, avi.. are just a container formats which contains multiple "data streams", which can be audio, video, subtitles (not all formats support this).
I don't know about your use case, but you don't need C code for what you do. You can just use gst-launch-1.0 tool which has its own GStreamer kind-of-scripting language.
For simplicity I will use debugging elements videotestsrc and audiotestsrc for simulating input (instead of actual camera etc).
gst-launch-1.0 -e videotestsrc ! x264enc ! mp4mux name=mux ! filesink location="bla.mp4" audiotestsrc ! lamemp3enc ! mux.
videotestsrc --> x264enc -----\
>---> mp4mux ---> filesink
audiotestsrc --> lamemp3enc --/
Explanation:
Videotestsrc generates raw video which is in GStreamer terms called "video/x-raw".
However mp4 cannot hold raw video, so we need to encode it with for example x264enc which makes our data "video/x-h264".
Then we can finally mux this into our mp4 with mp4mux element.
When we take a look into GStreamer docs using gst-inspect-1.0 mp4mux we see that this element supports various formats amongst which there is also video/x-h264.
The same thing we do with audio with either faac for AAC format or lamemp3enc for mp3.
With gst-launch-1.0 I did two tricks and one bonus trick:
ability to have separate branches in one line. This is achieved by just separating those branches with space instead of !
ability to make alias with name=mux and later on using it with adding dot right at the end of name like mux. . You can make up any name for that element you like.
Write EOS after hitting ctrl+c to stop the recording. This is achieved with parameter -e
Finally the output goes to filesink which just writes anything you give it to file.
Now for a homework you:
Use your elements for what you need - v4l2, alsasrc
Add queue elements to add buffering and thread separation

Timestamping error/or comptuer too slow with gstreamer/gstbasesink in Qt

I am building a simple video player in Qt using gstreamer-1.0. When I run it from Qt, or .exe in my pc, everything runs and works ok. But when I try it from another pc, it plays for some seconds that it skips some seconds/minutes and so on. I guess the problem is with sync, I have tried setting d3dvidesink property: sync=false, but is the same. I have read many similiar threads but none seems to help.
A lot of buffers are being dropped.
Additional debug info:
gstbasesink.c(2846): gst_base_sink_is_too_late ():
There may be a timestamping problem, or this computer is too slow.
I have tried setting different properties, but none helped. I have seen the following threads, but still the same problem:
Thread 1
Thread 2
Thread 3
On Thread 3 there is a suggestion setting "do-timestamp" property on appsrc to TRUE, but I use uridecodebin as source that has not a "do-timestamp" property.
My pipeline is as follows:
uridecodebin ! audioconvert ! volume ! autoaudiosink ! videoconvert ! gamma ! d3dvideosink
Thanks in Advance!
Here is some code from the elements creation/linking. Please comment if you need any other code.
// Create the elements
data.source = gst_element_factory_make ( "uridecodebin", "source" );
data.audio_convert = gst_element_factory_make ( "audioconvert", "audio_convert" );
data.volume = gst_element_factory_make ( "volume", "volume");
data.audio_sink = gst_element_factory_make ( "autoaudiosink", "audio_sink" );
data.video_convert = gst_element_factory_make ( "videoconvert", "video_convert" );
data.filter = gst_element_factory_make ( "gamma", "filter");
data.video_sink = gst_element_factory_make ( "d3dvideosink", "video_sink" );
// Create the empty pipeline
data.pipeline = gst_pipeline_new ("test-pipeline");
if (!data.pipeline || !data.source || !data.audio_convert || !data.volume || !data.audio_sink
|| !data.video_convert || !data.filter || !data.video_sink ) {
g_printerr ("Not all elements could be created.\n");}
return ;
}
// 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.audio_convert , data.volume, data.audio_sink,
data.video_convert, data.filter, data.video_sink, NULL);
if (!gst_element_link (data.audio_convert, data.volume)) {
g_printerr ("Elements AUDIO_CONVERT - VOLUME could not be linked.\n");
gst_object_unref (data.pipeline);
return ;
}
if (!gst_element_link (data.volume, data.audio_sink)) {
g_printerr ("Elements VOLUME - AUDIO_SINK could not be linked.\n");
gst_object_unref (data.pipeline);
return ;
}
if (!gst_element_link(data.video_convert, data.filter)) {
g_printerr("Elements VIDEO_CONVERT - FILTER could not be linked.\n");
gst_object_unref(data.pipeline);
return ;
}
if (!gst_element_link(data.filter, data.video_sink)) {
g_printerr("Elements FILTER - VIDEO_SINK could not be linked.\n");
gst_object_unref(data.pipeline);
return ;
}
When I open video:
// Set the URI to play
QString filePath = "file:///"+filename;
QByteArray ba = filePath.toLatin1();
const char *c_filePath = ba.data();
ret = gst_element_set_state (data.pipeline, GST_STATE_NULL);
gint64 max_lateness = 2000000; //2 milli sec
g_object_set (data.source, "uri", c_filePath, NULL);
// I have tried setting the following properties, but none helped
// g_object_set (data.source, "do-timestamp", true, NULL);
// g_object_set( data.video_sink, "sync", false, NULL);
// g_object_set( data.video_sink, "max-lateness", max_lateness, NULL);
qDebug() << &c_filePath;
// Link video_sink with playingWidget->winId()
gst_video_overlay_set_window_handle (GST_VIDEO_OVERLAY (data.video_sink), xwinid);
// 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) {
gst_element_set_state (data.pipeline, GST_STATE_NULL);
gst_object_unref (data.pipeline);
// Exit application
QTimer::singleShot(0, QApplication::activeWindow(), SLOT(quit()));}
data.playing = TRUE;
data.rate = 1.0;
// Iterate - gets the position and length every 200 msec
g_print ("Running...\n");
emit setMsg( "Running...\n" );
currFileName = filename;
timer->start(500);
Pad_added_handler:
void gst_pipeline::pad_added_handler(GstElement *src, GstPad *new_pad, CustomData *data)
{
GstPadLinkReturn ret;
GstCaps *new_pad_caps = NULL;
GstStructure *new_pad_struct = NULL;
const gchar *new_pad_type = NULL;
GstPad *sink_pad_audio = gst_element_get_static_pad (data->audio_queue, "sink");
GstPad *sink_pad_video = gst_element_get_static_pad (data->video_queue, "sink");
g_print ("Received new pad '%s' from '%s':\n", GST_PAD_NAME (new_pad), GST_ELEMENT_NAME (src));
// If our audio converter is already linked, we have nothing to do here
if (gst_pad_is_linked (sink_pad_audio))
{
g_print (" We have already linked sink_pad_audio. Ignoring.\n");
// goto exit;
}
// If our video converter is already linked, we have nothing to do here
if (gst_pad_is_linked (sink_pad_video))
{
g_print (" We have already linked sink_pad_video. Ignoring.\n");
// goto exit;
}
// Check the new pad's type
new_pad_caps = gst_pad_get_current_caps (new_pad); //gst_pad_get_caps
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, "audio/x-raw"))
{
// Attempt the link
ret = gst_pad_link (new_pad, sink_pad_audio);
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); }
}
else if (g_str_has_prefix (new_pad_type, "video/x-raw"))
{
// Attempt the link
ret = gst_pad_link (new_pad, sink_pad_video);
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); }
}
else
{
g_print (" It has type '%s' which is not audio/x-raw OR video/x-raw. Ignoring.\n", new_pad_type);
goto exit;
}
exit:
// Unreference the new pad's caps, if we got them
if (new_pad_caps != NULL)
{ gst_caps_unref (new_pad_caps); g_print("EXIT"); msg_STRING2 += "EXIT\n" ; }
// Unreference the sink pad
gst_object_unref (sink_pad_audio);
gst_object_unref (sink_pad_video);
}

Gstreamer source code doesnt work

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
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?
Well i figured it out. I don't know why but when i divided the source code into two separate ones and in one of them i included the code up until the UDPsink element and included the rest of the elements after that ( meaning udpsrc, rtpdepay and audiosink) in another source code file and compiled them separately in two separate Terminals it worked. I still don't know why it is like this , but i am happy that it works.
The sender and reciever are supposed to be two different processes, which is why it works when you use two terminals.
In your code, you're putting two different pipelines in the same pipeline element and setting it to playing.
This is not supported, you need to create a different pipeline for that.
pipeline1 = gst_pipeline_new ("src-pipeline");
pipeline2 = gst_pipeline_new ("sink-pipeline");

why is gstreamer caps is blocking pipeline

I am trying a basic pipeline below. If I run the pipeline without the caps function it passes straight through (assuming same I/O format). Once I add the caps the pre-roll starts but the video does not continue through to the sink output.
Am I coding this wrong?
thx
Art
#include <gst/gst.h>
#include <glib.h>
static void
on_pad_added (GstElement *element,
GstPad *pad,
gpointer data)
{
GstPad *sinkpad;
GstElement *decoder = (GstElement *) data;
g_print ("Dynamic pad created, linking out/in \n");
sinkpad = gst_element_get_static_pad (decoder, "sink");
gst_pad_link (pad, sinkpad);
gst_object_unref (sinkpad);
}
int
main (int argc,
char *argv[])
{
GMainLoop *loop;
gboolean link_ok;
GstElement *pipeline, *source, *decoder, *ffcs, *vidsc, *capsfout, *sink;
GstBus *bus;
GstCaps *caps;
/* Initialisation */
gst_init (&argc, &argv);
loop = g_main_loop_new (NULL, FALSE);
/* Create gstreamer elements */
pipeline = gst_pipeline_new ("video-player");
source = gst_element_factory_make ("filesrc", "file-source");
decoder = gst_element_factory_make ("decodebin2", "dec-bin2");
ffcs = gst_element_factory_make ("ffmpegcolorspace", "ffcs");
vidsc = gst_element_factory_make ("videoscale", "vidsc");
capsfout = gst_element_factory_make ("capsfilter", "capsfout");
sink = gst_element_factory_make ("filesink", "vidout");
if (!pipeline || !source || !decoder || !ffcs || !vidsc || !capsfout || !sink) {
g_printerr ("One element could not be created. Exiting.\n");
return -1;
}
/* we set the input/output filename to the source element */
g_object_set (G_OBJECT (source), "location", argv[1], NULL);
g_object_set (G_OBJECT (sink), "location", argv[2], NULL);
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
gst_bin_add_many (GST_BIN (pipeline),
source, decoder, ffcs, vidsc, capsfout, sink, NULL);
/* we link the elements together */
gst_element_link (source, decoder);
gst_element_link (decoder, ffcs);
gst_element_link (ffcs, vidsc);
caps = gst_caps_new_simple("video/x-raw-yuv",
"format", GST_TYPE_FOURCC, GST_MAKE_FOURCC ('I', '4', '2', '0'),
"width", G_TYPE_INT, 384,
"height", G_TYPE_INT, 216,
"framerate", GST_TYPE_FRACTION, 25, 1,
NULL);
link_ok = gst_element_link_filtered(vidsc,sink,caps);
gst_caps_unref (caps);
if (!link_ok) {
g_warning ("Failed to link vidsc to sink!");
}else{
g_print("seems ok, no error reported?\n");
}
/* Set the pipeline to "playing" state*/
g_print ("Now playing: %s\n", argv[1]);
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;
}
It may be late to answer but you maybe forgot to do the dynamic pad link between some elements and you never call the on_pad_added function. I had the same problem.
Have you tried to add it since then? For example this line to dynamically link two elements instead of the gst_element_link:
g_signal_connect (decoder, "pad-added", G_CALLBACK (on_pad_added), ffcs);
For what reasons do you need the capsfilter? Only constrain the properties you need. E.g. you set a framerate, but don't have a videorate element before the capsfilter. If you don't care about the framerate (e.g. just want to force a size) remove the framerate from the caps you set on capsfilter..