Following code is written to play a .wav file but it doesn't seem to work.
I would like to know if i am missing something in it.
Code:
#include <gst/gst.h>
#include <glib.h>
int main(int argc , char *argv[])
{
GMainLoop *loop;
GstElement *source,*audioparser,*sink,*pipeline;
GstBus *bus;
gst_init(&argc,&argv);
// create a pipeline
loop = g_main_loop_new (NULL, FALSE);
pipeline = gst_pipeline_new ("wav-player");
source = gst_element_factory_make("filesrc","file-source");
audioparser = gst_element_factory_make("wavparse","wav-parser");
sink = gst_element_factory_make("alsasink","sink1");
g_object_set (G_OBJECT (source), "location", argv[1], NULL);
gst_element_set_state (pipeline, GST_STATE_NULL);
// set location to current sourceg_object_set(G_OBJECT(source),"location",argv[1],NULL);
// add elements to bin
gst_bin_add_many(GST_BIN(pipeline),source,audioparser,sink,NULL);
gst_element_link_many(source,audioparser,sink,NULL);
// create bus
bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline));
gst_bus_add_watch (bus, bus_call, loop);
gst_object_unref (bus);
gst_element_set_state(pipeline, GST_STATE_PLAYING);
g_main_loop_run (loop);
return 1;
}
Please compile this using following command:
gcc -Wall $(pkg-config --cflags --libs gstreamer-0.10) wav.c -o wavparser
Thanks in advance
Just use playbin2 instead of a hand-crafted pipeline
that is, replace everything from "pipeline = gst_pipeline_new()" to "gst_element_link_many" by:
pipeline = gst_element_factory_make("playbin2", NULL);
g_object_set(pipeline, "uri", "file:///the/file/I/want.wav", NULL);
maybe playbin2 is exactly what you need, but answerring question: wavparse does not have static src-pad, so you must handle "pad-added" signal from this element on runtime. something like the code:
gst_bin_add_many (GST_BIN (pipeline), wavsrc, wavparse,audioconvert, audiosink, NULL);
g_object_set (G_OBJECT (wavsrc), "location", "sound.wav", NULL);
gst_element_link(wavsrc, wavparse);
gst_element_link(audioconvert, audiosink);
g_signal_connect (wavparse, "pad-added", G_CALLBACK (on_pad_added), audioconvert);
where: wavsrc is filesrc, wavparse is wavparse, audioconvert is audioconvert, audiosink is alsasink (i'm not sure alsasink works for you, so you can choose another)
void on_pad_added (GstElement *src_element, GstPad *src_pad, gpointer data)
{
g_print ("Linking dynamic pad...\n");
GstElement *sink_element = (GstElement *) data; // is audioconvert
GstPad *sink_pad = gst_element_get_static_pad (sink_element, "sink");
gst_pad_link (src_pad, sink_pad);
gst_object_unref (sink_pad);
src_element = NULL; // yup, i don't want "unused" warning here
}
Related
I made a .pcm audio file via the following command:
gst-launch-1.0 filesrc location=/home/pi/rawaudio/can-you-keep-a-secret.wav ! wavparse ! audioresample ! audioconvert ! audio/x-raw,format=S16BE,channels=1,rate=44100,layout=interleaved ! filesink location=/home/pi/rawaudio/test.pcm
I can play it with:
gst-launch-1.0 filesrc location=/home/pi/rawaudio/test.pcm ! audio/x-raw,format=S16BE,channels=1,rate=44100,layout=interleaved ! audioconvert ! audioresample ! alsasink
This is working perfectly. But now, I need to implement this into my c++ application. This is what I already have:
#include <string>
#include <stdio.h>
#include <gst/gst.h>
#include <gio/gio.h>
#include <boost/thread.hpp>
#define AUDIO_LOCATION "/home/pi/rawaudio/test.pcm"
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) {
gst_init(&argc, &argv);
GstElement *pipeline, *source, *parser, *sink, *convert;
GMainLoop *loop;
GstBus *bus;
guint bus_watch_id;
// loop
loop = g_main_loop_new(NULL, false);
// pipeline
pipeline = gst_pipeline_new("test_pipeline");
sink = gst_element_factory_make ("alsasink", "sink");
source = gst_element_factory_make("filesrc", "source");
g_object_set(G_OBJECT(source), "location", AUDIO_LOCATION, NULL);
//convert = gst_element_factory_make("audioconvert", "convert");
//parser = gst_element_factory_make("audioresample","parse");
GstPad *sourcepad, *sinkpad;
sourcepad = gst_element_get_static_pad(source, "src");
gst_pad_set_caps(sourcepad,
gst_caps_new_simple("audio/x-raw",
"rate", G_TYPE_INT, 44100,
"channels", G_TYPE_INT, 1,
"format", G_TYPE_STRING, "S16BE",
"layout", G_TYPE_STRING, "interleaved",
NULL));
gst_object_unref(sourcepad);
// bus
bus = gst_pipeline_get_bus(GST_PIPELINE (pipeline));
bus_watch_id = gst_bus_add_watch(bus, bus_call, loop);
gst_object_unref(bus);
// add elements into pipeline
gst_bin_add_many(GST_BIN(pipeline), source, sink, NULL);
// link source to decode
gst_element_link_many(source, sink, NULL);
// start playing
gst_element_set_state(GST_ELEMENT(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));
g_source_remove (bus_watch_id);
g_main_loop_unref(loop);
return 0;
}
But it is not working. I get
Error: The stream is in the wrong format
when running.
What am I missing? some elements like audioparse, audioconvert, audioresample ... I also tried inserting the audioconvert and audioresample after the source into the pipeline but then I get an internal data stream error exception.
Now I'm trying to play mp4 files.
After trying to play mp4 file to demux, a stream error occurs.
My code is below, please let me know if there are any problems.
Any advises? Thank you
error -> encountered a general stream error
int main(int argc, char* argv[]){
GMainLoop *loop;
GstElement *pipeline, *vsource, *demux, *vqueue, *ssource, *muxer, *filesink;
GstElement *vplay, *conv;
//GstBin *recording;
GstBus *bus;
GstPad *srcpad, *sinkpad;
guint bus_watch_id;
gst_init (&argc, &argv);
loop = g_main_loop_new (NULL, FALSE);
pipeline = gst_pipeline_new("Live Recording");
vsource = gst_element_factory_make("filesrc","v-file-source");
demux = gst_element_factory_make("qtdemux","v-file-demux");
vqueue = gst_element_factory_make("queue2","v-file-queue");
vplay = gst_element_factory_make("autovideosink","play");
conv = gst_element_factory_make("videoconvert","converter");
if(!pipeline)
g_print("no pipe\n");
if(!vsource)
g_print("no video source\n");
if(!demux)
g_print("no video demux\n");
if(!vqueue)
g_print("no video queue\n");
if(!vplay)
g_print("no video play\n");
if(!conv)
g_print("no conv");
g_object_set (G_OBJECT (vsource), "location", "./sample.mp4", 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),vsource,demux,vqueue, conv, vplay,NULL);
if(!gst_element_link(vsource, demux)){
g_print("no link");
}
if(!gst_element_link_many(vqueue, conv, vplay,NULL)){
g_print("no link many");
}
g_print("now play\n");
g_signal_connect (demux, "pad-added", G_CALLBACK (on_pad_added), vqueue);
gst_element_set_state(pipeline, GST_STATE_PLAYING);
g_main_loop_run(loop);
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref(GST_OBJECT(pipeline));
g_source_remove(bus_watch_id);
g_main_loop_unref(loop);
return 0;
}
I am trying to connect to a ubiquiti camera and successfully with the following command with gstreamer.
gst-launch-1.0 --gst-debug=4 rtspsrc location="rtsp://:554/live/ch00_0" ! rtph264depay ! h264parse ! openh264dec ! d3dvideosink
I look at the debug and it says could not link pads rtspsrc to rtph264depay.
Picture
but it gets the stream alright and can see the video. when I put this into a c project it says cannot link source to rtph264parse. I looked around it says to use a dynamic pad with the following code
static void on_pad_added (GstElement *element, GstPad *pad, gpointer data)
{
GstPad *sinkpad;
GstElement *decoder = (GstElement *) data;
/* We can now link this pad with the rtsp-decoder sink pad */
g_print ("Dynamic pad created, linking source/demuxer\n");
sinkpad = gst_element_get_static_pad (decoder, "sink");
gst_pad_link (pad, sinkpad);
gst_object_unref (sinkpad);
}
int main(int argc, char *argv[])
{
/* Initialize GStreamer */
gst_init(&argc,&argv);
/* Build Pipeline */
pipel.pipeline = gst_pipeline_new("My pipeline");
pipel.source = gst_element_factory_make ("rtspsrc","source");
g_object_set (G_OBJECT (pipel.source), "latency",2000,NULL);
pipel.rtppay = gst_element_factory_make( "rtph264depay", "depayl");
pipel.parse = gst_element_factory_make("h264parse","parse");
pipel.filter1 = gst_element_factory_make("capsfilter","filter");
pipel.decodebin = gst_element_factory_make ("openh264dec","decode");
pipel.sink = gst_element_factory_make("d3dvideosink","sink");
g_object_set (G_OBJECT (pipel.sink), "sync",FALSE,NULL);
//create_uri(url,url_size,ip_address,port);
g_object_set(GST_OBJECT(pipel.source),"location","rtsp://<IP>:554/live/ch00_0",NULL);
filtercaps = gst_caps_from_string("application/x-rtp");
g_object_set (G_OBJECT (pipel.filter1), "caps",filtercaps,NULL);
gst_caps_unref(filtercaps);
gst_bin_add_many (GST_BIN (pipel.pipeline),pipel.source
,pipel.rtppay
,pipel.parse
,pipel.decodebin
,pipel.sink
,NULL);
if(!gst_element_link(pipel.source,pipel.rtppay))
printf("\nFailed source to rtppay\n");
if(!gst_element_link_many(pipel.parse,pipel.decodebin,pipel.sink,NULL))
printf("\nFailed to link parse to sink");
g_signal_connect(pipel.rtppay, "pad-added", G_CALLBACK(on_pad_added), pipel.parse);
}
static void cb_new_rtspsrc_pad(GstElement *element,GstPad*pad,gpointer data)
{
gchar *name;
GstCaps * p_caps;
gchar * description;
GstElement *p_rtph264depay;
name = gst_pad_get_name(pad);
g_print("A new pad %s was created\n", name);
// here, you would setup a new pad link for the newly created pad
// sooo, now find that rtph264depay is needed and link them?
p_caps = gst_pad_get_pad_template_caps (pad);
description = gst_caps_to_string(p_caps);
printf("%s\n",p_caps,", ",description,"\n");
g_free(description);
p_rtph264depay = GST_ELEMENT(data);
// try to link the pads then ...
if(!gst_element_link_pads(element, name, p_rtph264depay, "sink"))
{
printf("Failed to link elements 3\n");
}
g_free(name);
}
/* ---------- Main --------------- */
int main(int argc, char *argv[])
{
/* Initialize GStreamer */
gst_init(&argc,&argv);
/* Build Pipeline */
pipel.pipeline = gst_pipeline_new("My pipeline");
creating_pipeline(ip_address,port);
pipel.source = gst_element_factory_make ("rtspsrc","source");
g_object_set (G_OBJECT (pipel.source), "latency",2000,NULL);
pipel.rtppay = gst_element_factory_make( "rtph264depay", "depayl");
pipel.parse = gst_element_factory_make("h264parse","parse");
pipel.filter1 = gst_element_factory_make("capsfilter","filter");
pipel.decodebin = gst_element_factory_make ("openh264dec","decode");
pipel.sink = gst_element_factory_make("d3dvideosink","sink");
g_object_set (G_OBJECT (pipel.sink), "sync",FALSE,NULL);
//create_uri(url,url_size,ip_address,port);
g_object_set(GST_OBJECT(pipel.source),"location","rtsp://<ip>:554/live/ch00_0",NULL);
filtercaps = gst_caps_from_string("application/x-rtp");
g_object_set (G_OBJECT (pipel.filter1), "caps",filtercaps,NULL);
gst_caps_unref(filtercaps);
gst_bin_add_many (GST_BIN (pipel.pipeline),pipel.source
,pipel.rtppay
,NULL);
// listen for newly created pads
g_signal_connect(pipel.source, "pad-added", G_CALLBACK(cb_new_rtspsrc_pad),pipel.rtppay);
gst_bin_add_many (GST_BIN (pipel.pipeline),pipel.parse,NULL);
if(!gst_element_link(pipel.rtppay,pipel.parse))
printf("\nNOPE\n");
gst_bin_add_many (GST_BIN (pipel.pipeline),pipel.decodebin
,pipel.sink
,NULL);
if(!gst_element_link_many(pipel.parse,pipel.decodebin,pipel.sink,NULL))
printf("\nFailed to link parse to sink");
g_signal_connect(pipel.rtppay, "pad-added", G_CALLBACK(on_pad_added), pipel.parse);
}
Works Now!
use cb_new_rtspsrc_pad to add the pad dynamically
add the parse to bin
link up between rtppay and parse
add the necessary elements the rest of the string to make it work.
I am using gst library to play multiple audio files and I have a problem:
Error: Internal data flow error
This is my code (without other demuxer and sink init):
GstElement *elm1, *elm2;
elm1 = gst_element_factory_make ("filesrc", "file1.ogg");
elm2 = gst_element_factory_make ("filesrc", "file2.ogg");
g_object_set (G_OBJECT (elm1), "location", "file1.ogg", NULL);
g_object_set (G_OBJECT (elm2), "location", "file2.ogg", NULL);
GstBus *bus;
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), elm1, elm2, demuxer, sink, NULL);
gst_element_link_many (elm1, elm2, demuxer);
g_signal_connect (demuxer, "pad-added", G_CALLBACK (on_pad_added), sink);
gst_element_set_state (pipeline, GST_STATE_PLAYING);
Did I make something wrong?
you need to listen for EOS for file-1 and then change the filesrc to file-2.
the following answer should help you -
'GStreamer dynamically change the filesrc location of a pipeline- No sound'
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..