Unable to play .wav file using gstreamer apis - gstreamer

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

GStreamer 1.0 - cannot play audio: Error: The stream is in the wrong format

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.

gstreamer : encountered a general stream error

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;
}

gstreamer 1.0 rtspsrc to rtph264depay cannot link

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.

Play multiple audio files with gst pipeline

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'

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..