<div dir="ltr"><div>Yeah, I see the line the creates decodebin element actually is commented out.</div><div>You should add it to the pipeline. decodebin decodes mp3 data to uncompressed audio.</div><div>audioresample, and most audiosinks, cannot take compressed data like mp3</div><div> </div><div>take a look at <a href="https://gstreamer.freedesktop.org/documentation/application-development/highlevel/playback-components.html">https://gstreamer.freedesktop.org/documentation/application-development/highlevel/playback-components.html</a></div><div> </div><div>Luca</div><div> </div></div><div class="gmail_extra"> <div class="gmail_quote">2018-03-16 14:13 GMT+01:00 Sujith reddy <<a href="mailto:Sujithreddy6192@gmail.com" target="_blank">Sujithreddy6192@gmail.com</a>>: <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">HI Luca, here in above code i didn't use decodebin element. ........................ i came to know that when i am giving mp3/m38u it is giving noise..then i rechecked the code i found out that i need to use decodebin element for decoding mp3. Now i tried with the below code .. it is saying *Elements could not be linked.* ///////////////////////////////////// /***************** gcc llll.c -o playback-tutorial-7 `pkg-config --cflags --libs gstreamer-1.0 gstreamer-audio-1.0 gstreamer-app-1.0` *******************/ #include <gstreamer-1.0/gst/gst.h> #include <gst/audio/audio.h> #include <string.h> #include <stdio.h> #define CHUNK_SIZE 4096 /* Amount of bytes we are sending in each buffer */ #define SAMPLE_RATE 48000 /* Samples per second we are sending */ /* Structure to contain all our information, so we can pass it to callbacks */ typedef struct _CustomData { GstElement *pipeline, *app_source, *tee, *audio_queue, *audio_convert1, *audio_resample, *audio_sink,*app_decode,*audio_decode; <div><div class="h5"> GstElement *app_queue, *audio_convert2, *app_sink; guint64 num_samples; /* Number of samples generated so far (for timestamp generation) */ // gfloat a, b, c, d; /* For waveform generation */ guint sourceid; /* To control the GSource */ FILE *fp,*fp1; GMainLoop *main_loop; /* GLib's Main Loop */ } CustomData; /* This method is called by the idle GSource in the mainloop, to feed CHUNK_SIZE bytes into appsrc. * The ide handler is added to the mainloop when appsrc requests us to start sending data (need-data signal) * and is removed when appsrc has enough data (enough-data signal). */ static gboolean push_data (CustomData *data) { GstBuffer *buffer; GstFlowReturn ret; int i,r; GstMapInfo map; gint num_samples = CHUNK_SIZE/2; /* Because each sample is 16 bits */ //gfloat freq; /* Create a new empty buffer */ buffer = gst_buffer_new_and_alloc (CHUNK_SIZE); /* Set its timestamp and duration */ GST_BUFFER_TIMESTAMP (buffer) = gst_util_uint64_scale (data->num_samples, GST_SECOND, SAMPLE_RATE); GST_BUFFER_DURATION (buffer) = gst_util_uint64_scale (CHUNK_SIZE, GST_SECOND, SAMPLE_RATE); /* Generate some psychodelic waveforms */ gst_buffer_map (buffer, &map, GST_MAP_WRITE); r=fread(map.data,2,CHUNK_SIZE/2,data->fp); gst_buffer_unmap (buffer, &map); data->num_samples += num_samples; while(r==NULL) gst_app_src_end_of_stream (data->app_source); /* Push the buffer into the appsrc */ g_signal_emit_by_name (data->app_source, "push-buffer", buffer, &ret); // gst_app_src_end_of_stream (data->app_source); //gst_app_src_push_buffer (data->app_source, buffer); /* Free the buffer now that we are done with it */ gst_buffer_unref (buffer); if (ret != GST_FLOW_OK) { /* We got some error, stop sending data */ return FALSE; } return TRUE; } /* This signal callback triggers when appsrc needs data. Here, we add an idle handler * to the mainloop to start pushing data into the appsrc */ 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); } } /* This callback triggers when appsrc has enough data and we can stop sending. * We remove the idle handler from the mainloop */ 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; } } /* The appsink has received a buffer */ static void new_sample (GstElement *sink, CustomData *data) { //printf("sujith1111111"); GstSample *sample; /////////////////////////////////////////////////////// GstBuffer *buffer; GstMapInfo map; g_signal_emit_by_name (data ->app_sink, "pull-sample", &sample,NULL); if (sample) { buffer = gst_sample_get_buffer (sample); gst_buffer_map (buffer, &map, GST_MAP_READ); g_print("\n here size=%d\n",map.size); fwrite(map.data,1,map.size,data->fp1); ///data is written to a file gst_buffer_unmap (buffer,&map); gst_sample_unref(sample); ///////////////////////////////////////////////// } } /* This function is called when an error message is posted on the bus */ static void error_cb (GstBus *bus, GstMessage *msg, CustomData *data) { GError *err; gchar *debug_info; /* Print error details on the screen */ 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); g_main_loop_quit (data->main_loop); } int main(int argc, char *argv[]) { CustomData data; GstPad *tee_audio_pad,*tee_app_pad; GstPad *queue_audio_pad, *queue_app_pad; GstAudioInfo info; GstCaps *audio_caps; GstBus *bus; /* Initialize cumstom data structure */ memset (&data, 0, sizeof (data)); </div></div>data.fp=fopen("/home/raghava/Documents/llll/songs/ChoosiChudangane.mp3","rb"); // data.fp= fopen("./Deviceconnected.raw","rb"); data.fp1 = fopen("1.raw","wb"); /* Initialize GStreamer */ gst_init (&argc, &argv); /* Create the elements */ data.app_source = gst_element_factory_make ("appsrc", "audio_source"); data.tee = gst_element_factory_make ("tee", "tee"); data.audio_queue = gst_element_factory_make ("queue", "audio_queue"); data.app_decode = gst_element_factory_make ("decodebin", "app_decode"); data.audio_convert1 = gst_element_factory_make ("audioconvert", "audio_convert1"); data.audio_resample = gst_element_factory_make ("audioresample", "audio_resample"); data.audio_sink = gst_element_factory_make ("autoaudiosink", "audio_sink"); data.app_queue = gst_element_factory_make ("queue", "app_queue"); data.audio_decode = gst_element_factory_make ("decodebin", "audio_decode"); data.audio_convert2 = gst_element_factory_make ("audioconvert", "audio_convert2"); data.app_sink = gst_element_factory_make ("appsink", "app_sink"); /* Create the empty pipeline */ data.pipeline = gst_pipeline_new ("test-pipeline"); if (!data.pipeline || !data.app_source || !data.tee || !data.audio_queue || !data.audio_convert1 || !data.audio_resample || !data.audio_sink || !data.audio_convert2 || !data.app_queue || !data.app_sink ||!data.audio_decode|| !data.app_decode ) // { g_printerr ("Not all elements could be created.\n"); return -1; } /* Configure appsrc */ gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16, SAMPLE_RATE, 1, NULL); audio_caps = gst_audio_info_to_caps (&info); g_object_set (data.app_source, "caps", audio_caps, "format", GST_FORMAT_TIME, NULL); //g_object_set (data.app_source, "format", GST_FORMAT_TIME, NULL); 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); /* Configure appsink */ g_object_set (data.app_sink, "emit-signals", TRUE, "caps", audio_caps, NULL); g_signal_connect (data.app_sink, "new-sample", G_CALLBACK (new_sample), &data); gst_caps_unref (audio_caps); // g_free (audio_caps_text); /* Link all elements that can be automatically linked because they have "Always" pads */ gst_bin_add_many (GST_BIN (data.pipeline), data.app_source, data.tee, data.audio_queue, data.audio_convert1, data.audio_resample, data.audio_sink, data.app_queue, data.audio_convert2,data.app_sink,data.audio_decode,data.app_decode, NULL);//,data.audio_decode,data.app_decode if (gst_element_link_many (data.app_source, data.tee, NULL) != TRUE || gst_element_link_many (data.audio_queue,data.audio_decode, data.audio_convert1, data.audio_resample, data.audio_sink, NULL) != TRUE || gst_element_link_many (data.app_queue,data.app_decode, <div><div class="h5">data.audio_convert2,data.app_sink, NULL) != TRUE )//,data.app_decode ,data.audio_decode { g_printerr ("Elements could not be linked.\n"); gst_object_unref (data.pipeline); return -1; } /* Manually link the Tee, which has "Request" pads */ tee_audio_pad = gst_element_get_request_pad (data.tee, "src_%u"); g_print ("Obtained request pad %s for audio branch.\n", gst_pad_get_name (tee_audio_pad)); queue_audio_pad = gst_element_get_static_pad (data.audio_queue, "sink"); tee_app_pad = gst_element_get_request_pad (data.tee, "src_%u"); g_print ("Obtained request pad %s for app branch.\n", gst_pad_get_name (tee_app_pad)); queue_app_pad = gst_element_get_static_pad (data.app_queue, "sink"); if (gst_pad_link (tee_audio_pad, queue_audio_pad) != GST_PAD_LINK_OK || gst_pad_link (tee_app_pad, queue_app_pad) != GST_PAD_LINK_OK) { g_printerr ("Tee could not be linked\n"); gst_object_unref (data.pipeline); return -1; } gst_object_unref (queue_audio_pad); gst_object_unref (queue_app_pad); /* Instruct the bus to emit signals for each received message, and connect to the interesting signals */ bus = gst_element_get_bus (data.pipeline); gst_bus_add_signal_watch (bus); g_signal_connect (G_OBJECT (bus), "message::error", (GCallback)error_cb, &data); gst_object_unref (bus); /* Start playing the pipeline */ gst_element_set_state (data.pipeline, GST_STATE_PLAYING); </div></div>/* sleep(6); gst_element_set_state (data.pipeline, GST_STATE_NULL); g_object_set (data.playbin, "uri", ", NULL);*/ /* Create a GLib Main Loop and set it to run */ int nstreams; g_object_get (data.pipeline, "n-audio", &nstreams, NULL); printf("nstreams =%d",nstreams); data.main_loop = g_main_loop_new (NULL, FALSE); g_main_loop_run (data.main_loop); /* Release the request pads from the Tee, and unref them */ gst_element_release_request_pad (data.tee, tee_audio_pad); gst_element_release_request_pad (data.tee, tee_app_pad); gst_object_unref (tee_audio_pad); gst_object_unref (tee_app_pad); /* Free resources */ gst_element_set_state (data.pipeline, GST_STATE_NULL); gst_object_unref (data.pipeline); return 0; } Thanks sujith <div class="HOEnZb"><div class="h5"> -- Sent from: <a href="http://gstreamer-devel.966125.n4.nabble.com/" target="_blank" rel="noreferrer">http://gstreamer-devel.966125.n4.nabble.com/</a> _______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a> <a href="https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" target="_blank" rel="noreferrer">https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> </div></div></blockquote></div> </div>