<div dir="ltr">Hi Everyone, I have two streaming pipelines and one rendering pipeline. First Streaming pipeline: (live source) uridecodebin uri=<StreamingURL> ! audioconvert ! audio/x-raw,format=S16LE,layout=interleaved,rate=48000 ! appsink name=sink1 g_signal_connect(g_sink1, "new-sample", G_CALLBACK(&onNewSampleFromSink), (gpointer)1); Second Streaming one (beep sound) from audiotestsrc: audiotestsrc is-live=true ! audioconvert ! audio/x-raw,format=S16LE,layout=interleaved,rate=48000 ! appsink name=sink2 g_signal_connect(g_sink2, "new-sample", G_CALLBACK(&onNewSampleFromSink), (gpointer)2); Rendering pipeline appsrc name=renderPipelineSource ! queue ! pulsesink or appsrc ! queue ! pulsesink name=renderPipelineSink With some user input i can control the appsink Callbacks and I decide which data to push to the rendering pipeline. (g_selectedSource = 1) GstFlowReturn onNewSampleFromSink(GstElement* const element, gpointer userData) { long val = (long)userData; GstSample* sample = gst_app_sink_pull_sample((GstAppSink*)element); if(g_selectedSource == val) { GstSample* copy = gst_sample_copy(sample); gst_app_src_push_sample((GstAppSrc*)data.renderPipelineSource, copy); gst_sample_unref(copy); } gst_sample_unref(sample); return GST_FLOW_OK; } For instance data from First Streaming pipeline are pushed to appsrc of rendering pipeline (live audio) data from Second streaming pipeline are not sent to the rendering pipeline. After some time I wanted to change the source from first to second. (g_selectedSource = 2) After changing data from Second Streaming pipeline are pushed to appsrc of rendering pipeline (Beep sound) data from First streaming pipeline are not sent to the rendering pipeline. The problem: Even after i Change the source to beep sound, I hear the live audio for even 4 to 5 seconds. I tried 1. gst_element_seek(renderPipeline, 1.0, GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH, GST_SEEK_TYPE_SET, 0, GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE); 2. gst_element_send_event(data.renderPipelineSource, gst_event_new_flush_start()); gst_element_send_event(data.renderPipelineSource, gst_event_new_flush_stop(TRUE)); 3. GstPad *sink_pad = gst_element_get_static_pad(data.renderPipelineSink, "sink"); GstEvent *flush_start_event = gst_event_new_flush_start(); GstEvent *flush_stop_event = gst_event_new_flush_stop(TRUE); /* Send flush-start event */ gst_pad_push_event(sink_pad, flush_start_event); gst_pad_send_event(sink_pad, flush_start_event); /* Wait for the pipeline to handle the flush-start event */ gst_element_get_state(data.renderPipeline, NULL, NULL, GST_CLOCK_TIME_NONE); /* Send flush-stop event */ gst_pad_push_event(sink_pad, flush_stop_event); gst_pad_send_event(sink_pad, flush_stop_event); gst_object_unref(sink_pad); 4. gst_element_set_state(data.renderPipeline, GST_STATE_PAUSED); gst_element_get_state(data.renderPipeline, NULL, NULL, GST_CLOCK_TIME_NONE); gst_element_set_state(data.renderPipeline, GST_STATE_PLAYING); 5. Setting properties of the src, queue and pulsesink of rendering pipeline g_object_set(G_OBJECT(data.renderPipelineSource), "max-bytes", 1024, NULL); // Set the maximum number of bytes g_object_set(G_OBJECT(data.renderPipelineSource), "max-time", 500000000, NULL); g_object_set(G_OBJECT(data.renderPipelineQueue), "max-size-bytes", 1024, NULL); // Set the maximum number of bytes g_object_set(G_OBJECT(data.renderPipelineQueue), "max-size-time", 500000000, NULL); g_object_set(G_OBJECT(data.renderPipelineSink), "blocksize", 1024, NULL); // Set the maximum number of bytes g_object_set(G_OBJECT(data.renderPipelineSink), "can-activate-pull", TRUE, NULL); // Set the maximum number of bytes None of the above methods helped. I do not want to set the renderPipeline to NULL then set it to Play state for internal reasons. Any helping leads are appreciated. Thanks in advance. Regards, Prakash </div>