<div style="line-height:1.7;color:#000000;font-size:14px;font-family:Arial"> <div id="isForwardContent"><div style="line-height:1.7;color:#000000;font-size:14px;font-family:Arial"><div>Hi,All</div><div> </div><div>I am pushing my PC camera and microphone to rtmp server by using lamemp3enc element. Currently,it can only push video h264 to rtmp server,but the time-delay is over 5 seconds,how to reduce it . After I add audio code into the project, VLC player failed to pull video/audio stream . Could you help me find the bug ? Code as following.</div><div style="line-height:1.7;color:#000000;font-size:14px;font-family:Arial"><div> </div><div>Thanks a lot!</div><div> </div><div> <div> </div><pre style="background-color:#ffffff;color:#000000;font-family:'Courier 10 Pitch';font-size:12.0pt;">#include <gst/gst.h> typedef struct _CustomData { GstElement *pipeline; GstElement *vsource; GstElement *vconvert; GstElement *x264enc; GstElement *h264parse; GstElement *vcapsfilter; GstElement *vsink; GstElement *asource; GstElement *aconvert; GstElement *amp3enc; GstElement *acapfilter; GstElement *ampegaudiopaser; GstElement *asink; GstElement *vqueue; GstElement *aqueue; GstElement *video_enc_queue; GstElement *audio_enc_queue; GstElement *flvmuxer; GstElement *rtmpsink; } CustomData; int main(int argc, char *argv[]) { CustomData data; GstBus *bus; GstMessage *msg; GstStateChangeReturn ret; gboolean terminate = FALSE; /* Initialize GStreamer */ //gst_init (&argc, &argv); gst_init(NULL, NULL); /* Create the elements */ //"video/x-raw, width=(int)160, height=(int)120, framerate=(fraction)30/1, format=I420, pixel-aspect-ratio=(fraction)1/1, interlace-mode=(string)progressive" data.vsource = gst_element_factory_make("v4l2src", "camera-source");//autovideosrc v4l2src data.vconvert = gst_element_factory_make("videoconvert", "convert"); data.vsink = gst_element_factory_make("ximagesink", "camera-sink"); data.vqueue = gst_element_factory_make("queue", "camera-queue"); data.x264enc = gst_element_factory_make("x264enc", "x264-encoder"); data.vcapsfilter = gst_element_factory_make("capsfilter", "video-filter"); data.h264parse = gst_element_factory_make("h264parse", "mux-video-parser");; data.asource = gst_element_factory_make("alsasrc", "audio-alsa-source"); data.aconvert = gst_element_factory_make("audioconvert", "audio-converter"); data.amp3enc = gst_element_factory_make("lamemp3enc", "mp3-encode"); data.acapfilter = gst_element_factory_make("capsfilter", "audio-capsfilter"); data.asink = gst_element_factory_make("filesink", "file-sink"); data.aqueue = gst_element_factory_make("queue", "audio-queue"); data.ampegaudiopaser = gst_element_factory_make("mpegaudioparse", "audio-paser"); data.flvmuxer = gst_element_factory_make("flvmux", "mux-flvmux"); data.rtmpsink = gst_element_factory_make("rtmpsink", "rtmp-sink"); data.video_enc_queue = gst_element_factory_make("queue", "video_enc_queue"); data.audio_enc_queue = gst_element_factory_make("queue", "audio_enc_queue"); g_object_set(G_OBJECT(data.rtmpsink), "location", "rtmp://localhost:1935/live/movie", NULL); g_object_set(G_OBJECT(data.flvmuxer), "streamable", true, NULL); /* Create the empty pipeline */ data.pipeline = gst_pipeline_new("push-pipeline"); if (!data.pipeline || !data.vsource || !data.vconvert || !data.vqueue || !data.x264enc || !data.vcapsfilter || !data.vsink || !data.asource || !data.aconvert || !data.amp3enc || !data.acapfilter || !data.asink || !data.aqueue) { g_printerr("Not all elements could be created.\n"); return -1; } gst_bin_add_many(GST_BIN(data.pipeline), data.asource, data.aqueue, data.aconvert, data.amp3enc, data.acapfilter, data.ampegaudiopaser, data.audio_enc_queue, NULL); gst_bin_add_many(GST_BIN(data.pipeline), data.vsource, data.vqueue, data.vconvert, data.x264enc, data.vcapsfilter, data.h264parse, data.video_enc_queue, NULL); gst_bin_add_many(GST_BIN(data.pipeline), data.rtmpsink, data.flvmuxer, NULL); //"video/x-h264, width=(int)640, height=(int)480, framerate=(fraction)30/1, stream-format=avc, alignment=au, profile=main" GstCaps *filtercaps = gst_caps_new_simple("video/x-h264", "stream-format", G_TYPE_STRING, "byte-stream",//avc,byte-stream "width", G_TYPE_INT, 320, "height", G_TYPE_INT, 240, "framerate", GST_TYPE_FRACTION, 30, 1, "alignment", G_TYPE_STRING, "au", "profile", G_TYPE_STRING, "main", NULL); g_object_set(G_OBJECT (data.vcapsfilter), "caps", filtercaps, NULL); bool v1 = gst_element_link(data.vsource, data.vconvert); if (!gst_element_link(data.vconvert, data.vqueue)) { g_printerr("Elements could not be linked.\n"); gst_object_unref(data.pipeline); return -1; } if (!gst_element_link(data.vqueue, data.x264enc)) { g_printerr("Elements could not be linked.\n"); gst_object_unref(data.pipeline); return -1; } if (!gst_element_link(data.x264enc, data.vcapsfilter)) { g_printerr("Elements could not be linked.\n"); gst_object_unref(data.pipeline); return -1; } if (!gst_element_link(data.vcapsfilter, data.h264parse)) { g_printerr("Elements could not be linked.\n"); gst_object_unref(data.pipeline); return -1; } if (!gst_element_link(data.h264parse, data.video_enc_queue)) { g_printerr("Elements could not be linked.\n"); gst_object_unref(data.pipeline); return -1; } //"audio/mpeg, mpegversion=1, layer=3, rate=(int)11025, channels=(int)2" GstCaps *fcaps = gst_caps_new_simple("audio/mpeg", "stream-format", G_TYPE_STRING, "byte-stream", "mpegversion", G_TYPE_INT, 1, "layer", G_TYPE_INT, 3, "rate", G_TYPE_INT, 11025,//44100 "channels", G_TYPE_INT, 2, NULL); g_object_set(G_OBJECT(data.acapfilter), "caps", fcaps, NULL); //g_object_set(G_OBJECT(data.asink), "location", "/home/greg/pre-push1.mp3", NULL); bool a0 = gst_element_link(data.asource, data.aqueue); bool a1 = gst_element_link(data.aqueue, data.aconvert); bool a2 = gst_element_link(data.aconvert, data.amp3enc); bool a3 = gst_element_link(data.amp3enc, data.acapfilter); bool aa = gst_element_link(data.acapfilter, data.ampegaudiopaser); bool xz = gst_element_link(data.ampegaudiopaser, data.audio_enc_queue); bool c1 = gst_element_link(data.audio_enc_queue, data.flvmuxer); bool c2 = gst_element_link(data.video_enc_queue, data.flvmuxer); bool c3 = gst_element_link(data.flvmuxer, data.rtmpsink); /* Set device=/dev/video0 */ g_object_set(data.vsource, "device", "/dev/video0", NULL); // g_object_set(G_OBJECT(data.amp3enc) , "target" , 1 , NULL) ; // g_object_set(G_OBJECT(data.amp3enc) , "cbr" , true , NULL) ; // CBR // g_object_set(G_OBJECT(data.amp3enc) , "bitrate" , 64 , NULL) ; // CBR ret = gst_element_set_state(data.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(data.pipeline); return -1; } /* Listen to the bus */ bus = gst_element_get_bus(data.pipeline); do { msg = gst_bus_timed_pop_filtered(bus, GST_CLOCK_TIME_NONE, (GstMessageType) (GST_MESSAGE_STATE_CHANGED | 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); terminate = TRUE; break; case GST_MESSAGE_EOS: g_print("End-Of-Stream reached.\n"); terminate = TRUE; break; case GST_MESSAGE_STATE_CHANGED: /* We are only interested in state-changed messages from the pipeline */ if (GST_MESSAGE_SRC (msg) == GST_OBJECT (data.pipeline)) { GstState old_state, new_state, pending_state; gst_message_parse_state_changed(msg, &old_state, &new_state, &pending_state); g_print("Pipeline state changed from %s to %s:\n", gst_element_state_get_name(old_state), gst_element_state_get_name(new_state)); } break; default: /* We should not reach here */ g_printerr("Unexpected message received.\n"); break; } gst_message_unref(msg); } } while (!terminate); /* Free resources */ gst_object_unref(bus); gst_element_set_state(data.pipeline, GST_STATE_NULL); gst_object_unref(data.pipeline); return 0; }</pre></div></div></div> </div></div>