我有一个GStreamer管道，可以从rtspsrc元素中提取视频。 rtspsrc元素连接到rtpjpegdepay元素。我希望能够动态更改RTSP URL。到目前为止，我一直在做的是：
1）取消链接rtspsrc与depay元素
2）使用新的RTSP URL
创建新的源元素3）并链接到depay元素。
我遇到的问题是新的RTSP源元素未正确链接到depay元素，从而导致段错误。我想帮助搞清楚如何动态更改rtspsrc URL（当管道仍在播放时）。
管道创建：
GstBus *bus;
guint busWatchId;
GstElement *src, *depay, *parser, *decoder, *vpe, *filter, *sink;
GstCaps *vpeCaps;
m_loop = g_main_loop_new(NULL, FALSE);
//create pipeline elements
m_cameraStream = gst_pipeline_new("display_pipeline");
src = gst_element_factory_make("rtspsrc", "rtspsrc");
depay = gst_element_factory_make("rtpjpegdepay", "depay");
parser = gst_element_factory_make("jpegparse", NULL);
decoder = gst_element_factory_make("ducatijpegdec", NULL);
vpe = gst_element_factory_make("vpe", NULL);
filter = gst_element_factory_make("capsfilter", NULL);
sink = gst_element_factory_make("waylandsink", NULL);
if(!(m_cameraStream || src || depay || parser || decoder || vpe || filter || sink)){
qFatal("could not create pipeline elements");
exit(1);
}
g_object_set(G_OBJECT(src), "location", "rtsp://192.168.50.29/av0_1", "latency", 0, NULL);
g_signal_connect(src, "pad-added", G_CALLBACK(on_rtsp_pad_added), depay);
//add src caps?
vpeCaps = gst_caps_from_string("video/x-raw, format=NV12, width=800, height=480"); //change this when Tomas' patch hits
if(!vpeCaps){
qFatal("cannot create caps");
exit(1);
}
g_object_set(G_OBJECT(filter), "caps", vpeCaps, NULL);
g_object_set(G_OBJECT(sink), "sync", false, NULL);
//add and link elements to create full pipeline
gst_bin_add_many(GST_BIN(m_cameraStream), src, depay, parser, decoder, vpe, sink, NULL);
if(!gst_element_link_many(depay, parser, decoder, vpe, sink, NULL)){
qFatal("cannot link elements");
exit(1);
}
gst_caps_unref(vpeCaps);
bus = gst_pipeline_get_bus(GST_PIPELINE(m_cameraStream));
busWatchId = gst_bus_add_watch(bus, GstBusFunc(bus_call), m_loop);
gst_object_unref(bus);
rtsp-＆gt; depay链接回调函数：
gchar *name;
GstElement *depay;
GstCaps *caps;
qDebug("on_rtsp_pad_added");
caps = gst_caps_from_string("application/x-rtp");
name = gst_pad_get_name(pad);
qDebug("on_rtsp_pad_added, rtspsrc pad name: %s", name);
depay = GST_ELEMENT(data);
if(!gst_element_link_pads_filtered(element, name, depay, "sink", caps)){
qFatal("pad_added: failed to link elements");
}
g_free(name);
gst_element_set_state(m_cameraStream, GST_STATE_PLAYING);
g_main_loop_run(m_loop);
来源变更功能：
qDebug("slot_changeSource");
//gst_element_set_state(m_cameraStream, GST_STATE_PAUSED); //GST_STATE_NULL: segfault in pad_added
//GST_STATE_PAUSED: pauses, never returns to playing or on_rtsp_pad_added
//GST_STATE_PLAYING(left playing): same as NULL
GstElement* rtspsrc = gst_bin_get_by_name(GST_BIN(m_cameraStream), "rtspsrc");
if(rtspsrc){
qDebug("rtspsrc found");
GstElement* depay = gst_bin_get_by_name(GST_BIN(m_cameraStream), "depay");
if(depay){
qDebug("depay found");
gst_element_unlink(rtspsrc, depay);
gst_bin_remove(GST_BIN(m_cameraStream), rtspsrc);
GstElement* newSource = gst_element_factory_make("rtspsrc", "rtspsrc");
g_object_set(G_OBJECT(newSource), "location", "rtsp://192.168.50.29/av0_1", "latency", 0, NULL);
g_signal_connect(newSource, "pad-added", G_CALLBACK(on_rtsp_pad_added), depay); //needed in the same way as the previous rtspsrc
gst_bin_add(GST_BIN(m_cameraStream), newSource);
gst_element_sync_state_with_parent(newSource);
//gst_element_set_state(m_cameraStream, GST_STATE_PLAYING);
}
gst_element_set_state(rtspsrc, GST_STATE_NULL);
gst_object_unref(rtspsrc);
}
我尝试过的其他事情：
1）探测rtsp元素的src pad，以确保元素中没有任何数据。这似乎是一个坏主意，因为此时rtsp元素将被新创建。
2）将管道设置为PAUSED或NULL，然后更改源元素。这导致管道永久停顿。
参考文献：
答案 0 :(得分：0)
好的，所以我相信我已经找到了答案，而且我会在这里发布这个，以便在这段时间内挽回任何绊倒。
答案是创建一对pad探测器来处理来自管道的清除数据。我通过创建两个pad探测器回调来完成此操作：一个用于捕获管道以开始刷新过程，另一个用于在刷新管道后处理rtspsrc元素的重新创建。第一个垫探针可以放在任何地方，所以我把它放在我的depay元素上。第二个pad探测器必须位于最后一个数据处理元素的源上。所以不是最后一个接收元素。对于上面的管道，这是＆＃39; vpe＆＃39;元件。
我这样做是通过将一个End of Stream（EOS）信号传递给depay元素，然后在vpe元素的src pad处进行一个pad probe回调，以便在退出VPE时捕获EOS。如果EOS进入waylandsink，管道就会关闭，你必须重新启动整个事情。
vpe = gst_bin_get_by_name(GST_BIN(data), "vpe");
srcPad = gst_element_get_static_pad(vpe, "src");
gst_pad_add_probe(srcPad, GST_PAD_PROBE_TYPE_BLOCK_DOWNSTREAM, event_probe, data, NULL);
//push EOS into the element, wait for the EOS to appear on the srcpad
depay = gst_bin_get_by_name(GST_BIN(data), "depay");
sinkPad = gst_element_get_static_pad(depay, "sink");
gst_pad_send_event(sinkPad, gst_event_new_eos());
return GST_PAD_PROBE_OK;
处理EOS的回调：
static GstPadProbeReturn event_probe(GstPad *pad, GstPadProbeInfo *info, gpointer data){
GstElement *rtspsrcOld, *rtspsrcNew, *depay;
qDebug("event_probe");
if(GST_EVENT_TYPE(GST_PAD_PROBE_INFO_DATA(info)) != GST_EVENT_EOS){
return GST_PAD_PROBE_PASS;
}
gst_pad_remove_probe(pad, GST_PAD_PROBE_INFO_ID(info));
rtspsrcOld = gst_bin_get_by_name(GST_BIN(data), "rtspsrc");
if(rtspsrcOld){
qDebug("found rtspsrcOld");
depay = gst_bin_get_by_name(GST_BIN(data), "depay");
gst_element_unlink(rtspsrcOld, depay);
gst_bin_remove(GST_BIN(data), rtspsrcOld); //remove old rtspsrc from pipeline, should unlink from depay automatically.
rtspsrcNew = gst_element_factory_make("rtspsrc", "rtspsrc");
g_object_set(rtspsrcNew, "location", NEW_URI, "latency", 0, NULL);
g_signal_connect(G_OBJECT(rtspsrcNew), "pad-added", G_CALLBACK(on_rtsp_pad_added), data);
gst_bin_add(GST_BIN(data), rtspsrcNew);
gst_element_set_state(GST_ELEMENT(data), GST_STATE_PLAYING);
return GST_PAD_PROBE_DROP;
}
return GST_PAD_PROBE_DROP;
}
答案 1 :(得分：0)
我试图做同样的事情。我刚开始使用 gstreamer。在理解了 T. Wallis 的意思后，我想用一个简单的管道来测试它。不幸的是，新的 rtspsrc 元素与管道的最终链接不起作用。但是，我认为错误在其他地方。我将再次通读代码并阅读gstreamer的动态管道操作程序。但我不确定，我是否能这么快找到错误。这是我的代码（这是我第一次发布堆栈溢出，对于潜在的 nogos 抱歉）：
#include <gst/gst.h>
#include <gst/gstpad.h>
#include <gst/rtsp/gstrtsp.h>
#include <unistd.h>
#include <time.h>
#include <stdbool.h>
typedef struct _CustomData {
GstElement *streaming_pipe;
GstElement *src;
GstElement *depay;
GstElement *decoder;
GstElement *sink;
GMainLoop *m_loop;
gboolean change_url;
gboolean url1;
clock_t startT;
} CustomData;
static void on_rtsp_pad_added(GstElement *element, GstPad *new_pad, CustomData *data){
gchar *name;
GstCaps *caps;
caps = gst_caps_from_string("application/x-rtp");
name = gst_pad_get_name(new_pad);
if(!gst_element_link_pads_filtered(element, name, data->depay, "sink", caps)){
g_print("\npad_added: failed to link elements"); //ERROR when linking the new rtspsrc after breaking up the pipeline
}
g_free(name);
data->startT = clock();
}
static GstPadProbeReturn event_probe(GstPad *pad, GstPadProbeInfo *info, CustomData *data){
GstElement *rtspsrcOld, *rtspsrcNew, *depay;
if(GST_EVENT_TYPE(GST_PAD_PROBE_INFO_DATA(info)) != GST_EVENT_EOS){
g_print("\n Not an EOS event; pass probe return");
return GST_PAD_PROBE_PASS;
}
gst_pad_remove_probe(pad, GST_PAD_PROBE_INFO_ID(info));
rtspsrcOld = gst_bin_get_by_name(GST_BIN(data->streaming_pipe), "rtspsrc");
if(rtspsrcOld){
depay = gst_bin_get_by_name(GST_BIN(data->streaming_pipe), "depay");
gst_element_unlink(rtspsrcOld, depay);
gst_bin_remove(GST_BIN(data->streaming_pipe), rtspsrcOld); //remove old rtspsrc from pipeline, should unlink from depay automatically.
rtspsrcNew = gst_element_factory_make("rtspsrc", "rtspsrc123");
g_object_set(rtspsrcNew, "location", "rtsp://xxxx/axis-media/media.amp?videocodec=h264&resolution=480x270", "latency", 0, NULL);
g_signal_connect(rtspsrcNew, "pad-added", G_CALLBACK(on_rtsp_pad_added), data);
gst_bin_add(GST_BIN(data->streaming_pipe), rtspsrcNew);
gst_element_set_state(GST_ELEMENT(data->streaming_pipe), GST_STATE_PLAYING);
g_print("\n set playing\n");
return GST_PAD_PROBE_DROP;
}
return GST_PAD_PROBE_DROP;
}
static GstPadProbeReturn cb_have_data (GstPad *pad, GstPadProbeInfo *info, CustomData *data) {
g_print("\nPROBE CALLBACK!");
g_print("Time: %f", ((double) (clock() - data->startT)) / CLOCKS_PER_SEC);
if(((double) (clock() - data->startT)) / CLOCKS_PER_SEC > 0.04){
data->change_url = true;
data->startT = clock();
}
if(data->change_url){
g_print("\nIF PROBE CALLBACK!");
GstPad *srcPad, *sinkPad;
srcPad = gst_element_get_static_pad(data->decoder, "src");
gst_pad_add_probe(srcPad, GST_PAD_PROBE_TYPE_BLOCK_DOWNSTREAM, event_probe, data, NULL);
//push EOS into the element, wait for the EOS to appear on the srcpad
sinkPad = gst_element_get_static_pad(data->depay, "sink");
gst_pad_send_event(sinkPad, gst_event_new_eos());
data->change_url = false;
}
return GST_PAD_PROBE_OK;
}
int main(int argc, char *argv[])
{
/* Initialize GStreamer */
gst_init (&argc, &argv);
CustomData data;
GstStateChangeReturn ret;
GstPad *pad;
data.m_loop = g_main_loop_new(NULL, FALSE);
//create pipeline elements
data.streaming_pipe = gst_pipeline_new("display_pipeline");
data.src = gst_element_factory_make("rtspsrc", "rtspsrc");
data.depay = gst_element_factory_make("rtph264depay", "depay");
data.decoder = gst_element_factory_make("avdec_h264", "decoder");
data.sink = gst_element_factory_make("autovideosink", NULL);
data.change_url = false;
data.url1 = false;
if(!(data.streaming_pipe || data.src || data.depay || data.decoder || data.sink)){
g_print("could not create pipeline elements");
exit(1);
}
g_object_set(G_OBJECT(data.src), "location", "rtsp://xxxx/axis-media/media.amp", "latency", 0, NULL);
g_signal_connect(data.src, "pad-added", G_CALLBACK(on_rtsp_pad_added), &data);
//add and link elements to create full pipeline
gst_bin_add_many(GST_BIN(data.streaming_pipe), data.src, data.depay, data.decoder, data.sink, NULL);
if(!gst_element_link_many(data.depay, data.decoder, data.sink, NULL)){
g_print("cannot link elements");
exit(1);
}
pad = gst_element_get_static_pad (data.depay, "src");
if(pad == NULL){
g_print("COULD NOT GET STATIC PAD");
}
gst_pad_add_probe (pad, GST_PAD_PROBE_TYPE_BUFFER,
(GstPadProbeCallback) cb_have_data, &data, NULL);
gst_object_unref (pad);
ret = gst_element_set_state (data.streaming_pipe, 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.streaming_pipe);
return -1;
}
g_main_loop_run (data.m_loop);
}