网络摄像机捕获
IP camera capture
我正在尝试捕获直接连接到 nVidia Jetson TK1 中的迷你 PCIe 双千兆位扩展卡的两个 IP 摄像机的流。
我通过以下命令使用 gstreamer 捕获了两个摄像头的流：
gst-launch-0.10 rtspsrc location=rtsp://admin:123456@192.168.0.123:554/mpeg4cif latency=0 ! decodebin ! ffmpegcolorspace ! autovideosink rtspsrc location=rtsp://admin:123456@192.168.2.254:554/mpeg4cif latency=0 ! decodebin ! ffmpegcolorspace ! autovideosink
每个摄像头显示一个 window，但在捕获开始时给出此输出：
WARNING: from element /GstPipeline:pipeline0/GstAutoVideoSink:autovideosink1/GstXvImageSink:autovideosink1-actual-sink-xvimage: A lot of buffers are being dropped.
Additional debug info:
gstbasesink.c(2875): gst_base_sink_is_too_late (): /GstPipeline:pipeline0/GstAutoVideoSink:autovideosink1/GstXvImageSink:autovideosink1-actual-sink-xvimage:
There may be a timestamping problem, or this computer is too slow.
---> TVMR: Video-conferencing detected !!!!!!!!!
流播放良好，"good" 摄像机之间也同步，但过了一会儿，突然其中一个摄像机停止，通常几秒钟后另一个也停止。使用像 Wireshark 这样的接口嗅探器，我可以检查 rtsp 数据包是否仍在从相机发送。
我的目的是使用这个相机将它们用作使用 openCV 的立体相机。我可以使用以下函数使用 OpenCV 捕获流：
camera[0].open("rtsp://admin:123456@192.168.2.254:554/mpeg4cif");//right
camera[1].open("rtsp://admin:123456@192.168.0.123:554/mpeg4cif");//left
它随机开始捕捉好坏，同步与否，延迟与否，但一段时间后无法使用捕获的图像，您可以在图像中观察到：
而运行openCV程序的输出通常是这样的：（我复制了最完整的一个）
[h264 @ 0x1b9580] slice type too large (2) at 0 23
[h264 @ 0x1b9580] decode_slice_header error
[h264 @ 0x1b1160] left block unavailable for requested intra mode at 0 6
[h264 @ 0x1b1160] error while decoding MB 0 6, bytestream (-1)
[h264 @ 0x1b1160] mmco: unref short failure
[h264 @ 0x1b9580] too many reference frames
[h264 @ 0x1b1160] pps_id (-1) out of range
使用的摄像头是两个SIP-1080J模块。
有谁知道如何使用 openCV 实现良好的捕获？首先摆脱那些 h264 消息并在程序执行时拥有稳定的图像。
如果没有，我如何使用 gstreamer 改进管道和缓冲区，以便在不突然停止流的情况下进行良好的捕获？。虽然我从来没有使用 gstreamer 通过 openCV 进行捕获，但也许有一天我会知道如何做并解决这个问题。
非常感谢。
经过几天的深入搜索和尝试，我直接打开使用gstreamer-0.10 API。首先，我通过 http://docs.gstreamer.com/pages/viewpage.action?pageId=327735
中的教程学习了如何使用它
对于大多数教程，您只需要安装 libgstreamer0.10-dev 和一些其他包。我全部安装了：
sudo apt-get install libgstreamer0*
然后将你想尝试的例子的代码复制到一个.c文件中，然后从终端在.c文件所在的文件夹中输入（在一些例子中你必须添加更多的库到pkg-config ):
gcc basic-tutorial-1.c $(pkg-config --cflags --libs gstreamer-0.10) -o basic-tutorial-1.c
在那之后我并没有感到迷茫，我开始尝试混合一些c 和c++ 代码。您可以使用适当的 g++ 命令或 CMakeLists.txt 或您想要的方式编译它...因为我正在使用 nVidia Jetson TK1 进行开发，所以我使用 Nsight Eclipse Edition 并且我需要配置项目属性正确使用 gstreamer-0.10 库和 openCV 库。
混合一些代码，最后我能够实时捕获我的两个 IP 摄像机的流，没有明显的延迟，没有任何帧中的错误解码，并且两个流同步。我还没有解决的唯一问题是获得彩色帧而不是灰度帧（我尝试使用其他 CV_ 值 "segmentation fault" 结果）：
v = Mat(Size(640, 360),CV_8U, (char*)GST_BUFFER_DATA(gstImageBuffer));
接下来是完整代码，我使用 gstreamer 捕获，将捕获转换为 openCV Mat 对象，然后显示它。该代码仅用于捕获一台 IP 摄像机。您可以复制同时捕获多个相机的对象和方法。
#include <opencv2/core/core.hpp>
#include <opencv2/contrib/contrib.hpp>
#include <opencv2/highgui/highgui.hpp>
#include <opencv2/imgproc/imgproc.hpp>
#include <opencv2/video/video.hpp>
#include <gst/gst.h>
#include <gst/app/gstappsink.h>
#include <gst/app/gstappbuffer.h>
#include <glib.h>
#define DEFAULT_LATENCY_MS 1
using namespace cv;
typedef struct _vc_cfg_data {
char server_ip_addr[100];
} vc_cfg_data;
typedef struct _vc_gst_data {
GMainLoop *loop;
GMainContext *context;
GstElement *pipeline;
GstElement *rtspsrc,*depayloader, *decoder, *converter, *sink;
GstPad *recv_rtp_src_pad;
} vc_gst_data;
typedef struct _vc_data {
vc_gst_data gst_data;
vc_cfg_data cfg;
} vc_data;
/* Global data */
vc_data app_data;
static void vc_pad_added_handler (GstElement *src, GstPad *new_pad, vc_data *data);
#define VC_CHECK_ELEMENT_ERROR(e, name) \
if (!e) { \
g_printerr ("Element %s could not be created. Exiting.\n", name); \
return -1; \
}
/*******************************************************************************
Gstreamer pipeline creation and init
*******************************************************************************/
int vc_gst_pipeline_init(vc_data *data)
{
GstStateChangeReturn ret;
// Template
GstPadTemplate* rtspsrc_pad_template;
// Create a new GMainLoop
data->gst_data.loop = g_main_loop_new (NULL, FALSE);
data->gst_data.context = g_main_loop_get_context(data->gst_data.loop);
// Create gstreamer elements
data->gst_data.pipeline = gst_pipeline_new ("videoclient");
VC_CHECK_ELEMENT_ERROR(data->gst_data.pipeline, "pipeline");
//RTP UDP Source - for received RTP messages
data->gst_data.rtspsrc = gst_element_factory_make ("rtspsrc", "rtspsrc");
VC_CHECK_ELEMENT_ERROR(data->gst_data.rtspsrc,"rtspsrc");
printf("URL: %s\n",data->cfg.server_ip_addr);
g_print ("Setting RTSP source properties: \n");
g_object_set (G_OBJECT (data->gst_data.rtspsrc), "location", data->cfg.server_ip_addr, "latency", DEFAULT_LATENCY_MS, NULL);
//RTP H.264 Depayloader
data->gst_data.depayloader = gst_element_factory_make ("rtph264depay","depayloader");
VC_CHECK_ELEMENT_ERROR(data->gst_data.depayloader,"rtph264depay");
//ffmpeg decoder
data->gst_data.decoder = gst_element_factory_make ("ffdec_h264", "decoder");
VC_CHECK_ELEMENT_ERROR(data->gst_data.decoder,"ffdec_h264");
data->gst_data.converter = gst_element_factory_make ("ffmpegcolorspace", "converter");
VC_CHECK_ELEMENT_ERROR(data->gst_data.converter,"ffmpegcolorspace");
// i.MX Video sink
data->gst_data.sink = gst_element_factory_make ("appsink", "sink");
VC_CHECK_ELEMENT_ERROR(data->gst_data.sink,"appsink");
gst_app_sink_set_max_buffers((GstAppSink*)data->gst_data.sink, 1);
gst_app_sink_set_drop ((GstAppSink*)data->gst_data.sink, TRUE);
g_object_set (G_OBJECT (data->gst_data.sink),"sync", FALSE, NULL);
//Request pads from rtpbin, starting with the RTP receive sink pad,
//This pad receives RTP data from the network (rtp-udpsrc).
rtspsrc_pad_template = gst_element_class_get_pad_template (GST_ELEMENT_GET_CLASS (data->gst_data.rtspsrc),"recv_rtp_src_0");
// Use the template to request the pad
data->gst_data.recv_rtp_src_pad = gst_element_request_pad (data->gst_data.rtspsrc, rtspsrc_pad_template,
"recv_rtp_src_0", NULL);
// Print the name for confirmation
g_print ("A new pad %s was created\n",
gst_pad_get_name (data->gst_data.recv_rtp_src_pad));
// Add elements into the pipeline
g_print(" Adding elements to pipeline...\n");
gst_bin_add_many (GST_BIN (data->gst_data.pipeline),
data->gst_data.rtspsrc,
data->gst_data.depayloader,
data->gst_data.decoder,
data->gst_data.converter,
data->gst_data.sink,
NULL);
// Link some of the elements together
g_print(" Linking some elements ...\n");
if(!gst_element_link_many (data->gst_data.depayloader, data->gst_data.decoder, data->gst_data.converter, data->gst_data.sink, NULL))
g_print("Error: could not link all elements\n");
// Connect to the pad-added signal for the rtpbin. This allows us to link
//the dynamic RTP source pad to the depayloader when it is created.
if(!g_signal_connect (data->gst_data.rtspsrc, "pad-added",
G_CALLBACK (vc_pad_added_handler), data))
g_print("Error: could not add signal handler\n");
// Set the pipeline to "playing" state
g_print ("Now playing A\n");
ret = gst_element_set_state (data->gst_data.pipeline, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE) {
g_printerr ("Unable to set the pipeline A to the playing state.\n");
gst_object_unref (data->gst_data.pipeline);
return -1;
}
return 0;
}
static void vc_pad_added_handler (GstElement *src, GstPad *new_pad, vc_data *data) {
GstPad *sink_pad = gst_element_get_static_pad (data->gst_data.depayloader, "sink");
GstPadLinkReturn ret;
GstCaps *new_pad_caps = NULL;
GstStructure *new_pad_struct = NULL;
const gchar *new_pad_type = NULL;
g_print ("Received new pad '%s' from '%s':\n", GST_PAD_NAME (new_pad), GST_ELEMENT_NAME (src));
/* Check the new pad's name */
if (!g_str_has_prefix (GST_PAD_NAME (new_pad), "recv_rtp_src_")) {
g_print (" It is not the right pad. Need recv_rtp_src_. Ignoring.\n");
goto exit;
}
/* If our converter is already linked, we have nothing to do here */
if (gst_pad_is_linked (sink_pad)) {
g_print (" Sink pad from %s already linked. Ignoring.\n", GST_ELEMENT_NAME (src));
goto exit;
}
/* Check the new pad's type */
new_pad_caps = gst_pad_get_caps (new_pad);
new_pad_struct = gst_caps_get_structure (new_pad_caps, 0);
new_pad_type = gst_structure_get_name (new_pad_struct);
/* Attempt the link */
ret = gst_pad_link (new_pad, sink_pad);
if (GST_PAD_LINK_FAILED (ret)) {
g_print (" Type is '%s' but link failed.\n", new_pad_type);
} else {
g_print (" Link succeeded (type '%s').\n", new_pad_type);
}
exit:
/* Unreference the new pad's caps, if we got them */
if (new_pad_caps != NULL)
gst_caps_unref (new_pad_caps);
/* Unreference the sink pad */
gst_object_unref (sink_pad);
}
int vc_gst_pipeline_clean(vc_data *data) {
GstStateChangeReturn ret;
GstStateChangeReturn ret2;
/* Cleanup Gstreamer */
if(!data->gst_data.pipeline)
return 0;
/* Send the main loop a quit signal */
g_main_loop_quit(data->gst_data.loop);
g_main_loop_unref(data->gst_data.loop);
ret = gst_element_set_state (data->gst_data.pipeline, GST_STATE_NULL);
if (ret == GST_STATE_CHANGE_FAILURE) {
g_printerr ("Unable to set the pipeline A to the NULL state.\n");
gst_object_unref (data->gst_data.pipeline);
return -1;
}
g_print ("Deleting pipeline\n");
gst_object_unref (GST_OBJECT (data->gst_data.pipeline));
/* Zero out the structure */
memset(&data->gst_data, 0, sizeof(vc_gst_data));
return 0;
}
void handleKey(char key)
{
switch (key)
{
case 27:
break;
}
}
int vc_mainloop(vc_data* data)
{
GstBuffer *gstImageBuffer;
Mat v;
namedWindow("view",WINDOW_NORMAL);
while (1) {
gstImageBuffer = gst_app_sink_pull_buffer((GstAppSink*)data->gst_data.sink);
if (gstImageBuffer != NULL )
{
v = Mat(Size(640, 360),CV_8U, (char*)GST_BUFFER_DATA(gstImageBuffer));
imshow("view", v);
handleKey((char)waitKey(3));
gst_buffer_unref(gstImageBuffer);
}else{
g_print("gsink buffer didn't return buffer.");
}
}
return 0;
}
int main (int argc, char *argv[])
{
setenv("DISPLAY", ":0", 0);
strcpy(app_data.cfg.server_ip_addr, "rtsp://admin:123456@192.168.0.123:554/mpeg4cif");
gst_init (&argc, &argv);
if(vc_gst_pipeline_init(&app_data) == -1) {
printf("Gstreamer pipeline creation and init failed\n");
goto cleanup;
}
vc_mainloop(&app_data);
printf ("Returned, stopping playback\n");
cleanup:
return vc_gst_pipeline_clean(&app_data);
return 0;
}
希望对您有所帮助！！ ;)
uri = 'rtsp://admin:123456@192.168.0.123:554/mpeg4cif'
gst_str = ("rtspsrc location={} latency={} ! rtph264depay ! h264parse ! omxh264dec ! nvvidconv ! video/x-raw, width=(int){}, height=(int){}, format=(string)BGRx ! videoconvert ! appsink sync=false").format(uri, 200, 3072, 2048)
cap= cv2.VideoCapture(gst_str,cv2.CAP_GSTREAMER)
while(True):
_,frame = cap.read()
if frame is None:
break
cv2.imshow("",frame)
cv2.waitKey(0)
cap.release()
cv2.destroyAllWindows()
我正在尝试捕获直接连接到 nVidia Jetson TK1 中的迷你 PCIe 双千兆位扩展卡的两个 IP 摄像机的流。
我通过以下命令使用 gstreamer 捕获了两个摄像头的流：
gst-launch-0.10 rtspsrc location=rtsp://admin:123456@192.168.0.123:554/mpeg4cif latency=0 ! decodebin ! ffmpegcolorspace ! autovideosink rtspsrc location=rtsp://admin:123456@192.168.2.254:554/mpeg4cif latency=0 ! decodebin ! ffmpegcolorspace ! autovideosink
每个摄像头显示一个 window，但在捕获开始时给出此输出：
WARNING: from element /GstPipeline:pipeline0/GstAutoVideoSink:autovideosink1/GstXvImageSink:autovideosink1-actual-sink-xvimage: A lot of buffers are being dropped.
Additional debug info:
gstbasesink.c(2875): gst_base_sink_is_too_late (): /GstPipeline:pipeline0/GstAutoVideoSink:autovideosink1/GstXvImageSink:autovideosink1-actual-sink-xvimage:
There may be a timestamping problem, or this computer is too slow.
---> TVMR: Video-conferencing detected !!!!!!!!!
流播放良好，"good" 摄像机之间也同步，但过了一会儿，突然其中一个摄像机停止，通常几秒钟后另一个也停止。使用像 Wireshark 这样的接口嗅探器，我可以检查 rtsp 数据包是否仍在从相机发送。
我的目的是使用这个相机将它们用作使用 openCV 的立体相机。我可以使用以下函数使用 OpenCV 捕获流：
camera[0].open("rtsp://admin:123456@192.168.2.254:554/mpeg4cif");//right
camera[1].open("rtsp://admin:123456@192.168.0.123:554/mpeg4cif");//left
它随机开始捕捉好坏，同步与否，延迟与否，但一段时间后无法使用捕获的图像，您可以在图像中观察到：
而运行openCV程序的输出通常是这样的：（我复制了最完整的一个）
[h264 @ 0x1b9580] slice type too large (2) at 0 23
[h264 @ 0x1b9580] decode_slice_header error
[h264 @ 0x1b1160] left block unavailable for requested intra mode at 0 6
[h264 @ 0x1b1160] error while decoding MB 0 6, bytestream (-1)
[h264 @ 0x1b1160] mmco: unref short failure
[h264 @ 0x1b9580] too many reference frames
[h264 @ 0x1b1160] pps_id (-1) out of range
使用的摄像头是两个SIP-1080J模块。
有谁知道如何使用 openCV 实现良好的捕获？首先摆脱那些 h264 消息并在程序执行时拥有稳定的图像。
如果没有，我如何使用 gstreamer 改进管道和缓冲区，以便在不突然停止流的情况下进行良好的捕获？。虽然我从来没有使用 gstreamer 通过 openCV 进行捕获，但也许有一天我会知道如何做并解决这个问题。
非常感谢。
经过几天的深入搜索和尝试，我直接打开使用gstreamer-0.10 API。首先，我通过 http://docs.gstreamer.com/pages/viewpage.action?pageId=327735
中的教程学习了如何使用它对于大多数教程，您只需要安装 libgstreamer0.10-dev 和一些其他包。我全部安装了：
sudo apt-get install libgstreamer0*
然后将你想尝试的例子的代码复制到一个.c文件中，然后从终端在.c文件所在的文件夹中输入（在一些例子中你必须添加更多的库到pkg-config ):
gcc basic-tutorial-1.c $(pkg-config --cflags --libs gstreamer-0.10) -o basic-tutorial-1.c
在那之后我并没有感到迷茫，我开始尝试混合一些c 和c++ 代码。您可以使用适当的 g++ 命令或 CMakeLists.txt 或您想要的方式编译它...因为我正在使用 nVidia Jetson TK1 进行开发，所以我使用 Nsight Eclipse Edition 并且我需要配置项目属性正确使用 gstreamer-0.10 库和 openCV 库。
混合一些代码，最后我能够实时捕获我的两个 IP 摄像机的流，没有明显的延迟，没有任何帧中的错误解码，并且两个流同步。我还没有解决的唯一问题是获得彩色帧而不是灰度帧（我尝试使用其他 CV_ 值 "segmentation fault" 结果）：
v = Mat(Size(640, 360),CV_8U, (char*)GST_BUFFER_DATA(gstImageBuffer));
接下来是完整代码，我使用 gstreamer 捕获，将捕获转换为 openCV Mat 对象，然后显示它。该代码仅用于捕获一台 IP 摄像机。您可以复制同时捕获多个相机的对象和方法。
#include <opencv2/core/core.hpp>
#include <opencv2/contrib/contrib.hpp>
#include <opencv2/highgui/highgui.hpp>
#include <opencv2/imgproc/imgproc.hpp>
#include <opencv2/video/video.hpp>
#include <gst/gst.h>
#include <gst/app/gstappsink.h>
#include <gst/app/gstappbuffer.h>
#include <glib.h>
#define DEFAULT_LATENCY_MS 1
using namespace cv;
typedef struct _vc_cfg_data {
char server_ip_addr[100];
} vc_cfg_data;
typedef struct _vc_gst_data {
GMainLoop *loop;
GMainContext *context;
GstElement *pipeline;
GstElement *rtspsrc,*depayloader, *decoder, *converter, *sink;
GstPad *recv_rtp_src_pad;
} vc_gst_data;
typedef struct _vc_data {
vc_gst_data gst_data;
vc_cfg_data cfg;
} vc_data;
/* Global data */
vc_data app_data;
static void vc_pad_added_handler (GstElement *src, GstPad *new_pad, vc_data *data);
#define VC_CHECK_ELEMENT_ERROR(e, name) \
if (!e) { \
g_printerr ("Element %s could not be created. Exiting.\n", name); \
return -1; \
}
/*******************************************************************************
Gstreamer pipeline creation and init
*******************************************************************************/
int vc_gst_pipeline_init(vc_data *data)
{
GstStateChangeReturn ret;
// Template
GstPadTemplate* rtspsrc_pad_template;
// Create a new GMainLoop
data->gst_data.loop = g_main_loop_new (NULL, FALSE);
data->gst_data.context = g_main_loop_get_context(data->gst_data.loop);
// Create gstreamer elements
data->gst_data.pipeline = gst_pipeline_new ("videoclient");
VC_CHECK_ELEMENT_ERROR(data->gst_data.pipeline, "pipeline");
//RTP UDP Source - for received RTP messages
data->gst_data.rtspsrc = gst_element_factory_make ("rtspsrc", "rtspsrc");
VC_CHECK_ELEMENT_ERROR(data->gst_data.rtspsrc,"rtspsrc");
printf("URL: %s\n",data->cfg.server_ip_addr);
g_print ("Setting RTSP source properties: \n");
g_object_set (G_OBJECT (data->gst_data.rtspsrc), "location", data->cfg.server_ip_addr, "latency", DEFAULT_LATENCY_MS, NULL);
//RTP H.264 Depayloader
data->gst_data.depayloader = gst_element_factory_make ("rtph264depay","depayloader");
VC_CHECK_ELEMENT_ERROR(data->gst_data.depayloader,"rtph264depay");
//ffmpeg decoder
data->gst_data.decoder = gst_element_factory_make ("ffdec_h264", "decoder");
VC_CHECK_ELEMENT_ERROR(data->gst_data.decoder,"ffdec_h264");
data->gst_data.converter = gst_element_factory_make ("ffmpegcolorspace", "converter");
VC_CHECK_ELEMENT_ERROR(data->gst_data.converter,"ffmpegcolorspace");
// i.MX Video sink
data->gst_data.sink = gst_element_factory_make ("appsink", "sink");
VC_CHECK_ELEMENT_ERROR(data->gst_data.sink,"appsink");
gst_app_sink_set_max_buffers((GstAppSink*)data->gst_data.sink, 1);
gst_app_sink_set_drop ((GstAppSink*)data->gst_data.sink, TRUE);
g_object_set (G_OBJECT (data->gst_data.sink),"sync", FALSE, NULL);
//Request pads from rtpbin, starting with the RTP receive sink pad,
//This pad receives RTP data from the network (rtp-udpsrc).
rtspsrc_pad_template = gst_element_class_get_pad_template (GST_ELEMENT_GET_CLASS (data->gst_data.rtspsrc),"recv_rtp_src_0");
// Use the template to request the pad
data->gst_data.recv_rtp_src_pad = gst_element_request_pad (data->gst_data.rtspsrc, rtspsrc_pad_template,
"recv_rtp_src_0", NULL);
// Print the name for confirmation
g_print ("A new pad %s was created\n",
gst_pad_get_name (data->gst_data.recv_rtp_src_pad));
// Add elements into the pipeline
g_print(" Adding elements to pipeline...\n");
gst_bin_add_many (GST_BIN (data->gst_data.pipeline),
data->gst_data.rtspsrc,
data->gst_data.depayloader,
data->gst_data.decoder,
data->gst_data.converter,
data->gst_data.sink,
NULL);
// Link some of the elements together
g_print(" Linking some elements ...\n");
if(!gst_element_link_many (data->gst_data.depayloader, data->gst_data.decoder, data->gst_data.converter, data->gst_data.sink, NULL))
g_print("Error: could not link all elements\n");
// Connect to the pad-added signal for the rtpbin. This allows us to link
//the dynamic RTP source pad to the depayloader when it is created.
if(!g_signal_connect (data->gst_data.rtspsrc, "pad-added",
G_CALLBACK (vc_pad_added_handler), data))
g_print("Error: could not add signal handler\n");
// Set the pipeline to "playing" state
g_print ("Now playing A\n");
ret = gst_element_set_state (data->gst_data.pipeline, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE) {
g_printerr ("Unable to set the pipeline A to the playing state.\n");
gst_object_unref (data->gst_data.pipeline);
return -1;
}
return 0;
}
static void vc_pad_added_handler (GstElement *src, GstPad *new_pad, vc_data *data) {
GstPad *sink_pad = gst_element_get_static_pad (data->gst_data.depayloader, "sink");
GstPadLinkReturn ret;
GstCaps *new_pad_caps = NULL;
GstStructure *new_pad_struct = NULL;
const gchar *new_pad_type = NULL;
g_print ("Received new pad '%s' from '%s':\n", GST_PAD_NAME (new_pad), GST_ELEMENT_NAME (src));
/* Check the new pad's name */
if (!g_str_has_prefix (GST_PAD_NAME (new_pad), "recv_rtp_src_")) {
g_print (" It is not the right pad. Need recv_rtp_src_. Ignoring.\n");
goto exit;
}
/* If our converter is already linked, we have nothing to do here */
if (gst_pad_is_linked (sink_pad)) {
g_print (" Sink pad from %s already linked. Ignoring.\n", GST_ELEMENT_NAME (src));
goto exit;
}
/* Check the new pad's type */
new_pad_caps = gst_pad_get_caps (new_pad);
new_pad_struct = gst_caps_get_structure (new_pad_caps, 0);
new_pad_type = gst_structure_get_name (new_pad_struct);
/* Attempt the link */
ret = gst_pad_link (new_pad, sink_pad);
if (GST_PAD_LINK_FAILED (ret)) {
g_print (" Type is '%s' but link failed.\n", new_pad_type);
} else {
g_print (" Link succeeded (type '%s').\n", new_pad_type);
}
exit:
/* Unreference the new pad's caps, if we got them */
if (new_pad_caps != NULL)
gst_caps_unref (new_pad_caps);
/* Unreference the sink pad */
gst_object_unref (sink_pad);
}
int vc_gst_pipeline_clean(vc_data *data) {
GstStateChangeReturn ret;
GstStateChangeReturn ret2;
/* Cleanup Gstreamer */
if(!data->gst_data.pipeline)
return 0;
/* Send the main loop a quit signal */
g_main_loop_quit(data->gst_data.loop);
g_main_loop_unref(data->gst_data.loop);
ret = gst_element_set_state (data->gst_data.pipeline, GST_STATE_NULL);
if (ret == GST_STATE_CHANGE_FAILURE) {
g_printerr ("Unable to set the pipeline A to the NULL state.\n");
gst_object_unref (data->gst_data.pipeline);
return -1;
}
g_print ("Deleting pipeline\n");
gst_object_unref (GST_OBJECT (data->gst_data.pipeline));
/* Zero out the structure */
memset(&data->gst_data, 0, sizeof(vc_gst_data));
return 0;
}
void handleKey(char key)
{
switch (key)
{
case 27:
break;
}
}
int vc_mainloop(vc_data* data)
{
GstBuffer *gstImageBuffer;
Mat v;
namedWindow("view",WINDOW_NORMAL);
while (1) {
gstImageBuffer = gst_app_sink_pull_buffer((GstAppSink*)data->gst_data.sink);
if (gstImageBuffer != NULL )
{
v = Mat(Size(640, 360),CV_8U, (char*)GST_BUFFER_DATA(gstImageBuffer));
imshow("view", v);
handleKey((char)waitKey(3));
gst_buffer_unref(gstImageBuffer);
}else{
g_print("gsink buffer didn't return buffer.");
}
}
return 0;
}
int main (int argc, char *argv[])
{
setenv("DISPLAY", ":0", 0);
strcpy(app_data.cfg.server_ip_addr, "rtsp://admin:123456@192.168.0.123:554/mpeg4cif");
gst_init (&argc, &argv);
if(vc_gst_pipeline_init(&app_data) == -1) {
printf("Gstreamer pipeline creation and init failed\n");
goto cleanup;
}
vc_mainloop(&app_data);
printf ("Returned, stopping playback\n");
cleanup:
return vc_gst_pipeline_clean(&app_data);
return 0;
}
希望对您有所帮助！！ ;)
uri = 'rtsp://admin:123456@192.168.0.123:554/mpeg4cif'
gst_str = ("rtspsrc location={} latency={} ! rtph264depay ! h264parse ! omxh264dec ! nvvidconv ! video/x-raw, width=(int){}, height=(int){}, format=(string)BGRx ! videoconvert ! appsink sync=false").format(uri, 200, 3072, 2048)
cap= cv2.VideoCapture(gst_str,cv2.CAP_GSTREAMER)
while(True):
_,frame = cap.read()
if frame is None:
break
cv2.imshow("",frame)
cv2.waitKey(0)
cap.release()
cv2.destroyAllWindows()