2015-02-23 230 views
1

我试图捕获直接连接到nVidia Jetson TK1中的迷你PCIe双千兆位扩展卡的两个IP摄像头的流。IP摄像头捕获

我取得捕捉到的使用GStreamer的下一个命令两个相机流:

gst-launch-0.10 rtspsrc location=rtsp://admin:[email protected]:554/mpeg4cif latency=0 ! decodebin ! ffmpegcolorspace ! autovideosink rtspsrc location=rtsp://admin:[email protected]:554/mpeg4cif latency=0 ! decodebin ! ffmpegcolorspace ! autovideosink 

它显示每台摄像机一个窗口,但给出了这样的输出只是当拍摄开始:

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,相机之间也有“良好”的同步,但过了一段时间,突然有一台摄像机停下来,通常几秒钟后另一台也停下来。使用像Wireshark这样的接口snifer,我可以检查rtsp数据包是否仍然从相机发送。

我的目的是使用这台摄像机作为使用openCV的立体相机。我能够与OpenCV中捕捉到流具有以下功能:

camera[0].open("rtsp://admin:[email protected]:554/mpeg4cif");//right 
camera[1].open("rtsp://admin:[email protected]:554/mpeg4cif");//left 

它randomnly开始捕获好坏,同步或不同步,以延迟或没有,但一段时间后是无法使用拍摄的图像你可以在图像中观察到:

enter image description here

和输出运行OpenCV的程序时通常是这样的:(我抄了最完整的一个)

[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 

用过的相机是两个SIP-1080J模块。

任何人都知道如何使用openCV实现良好的捕捉?首先摆脱这些h264消息,并在程序执行时获得稳定的图像。

如果不是,我该如何使用gstreamer改进管道和缓冲区,以便在突然停止流的情况下获得良好的捕获?虽然我从来没有通过使用gstreamer的openCV获取,也许有一天我会知道如何去做并解决这个问题。

非常感谢。

+0

尝试播放编码器参数 - 首先尝试使用基线配置文件,降低比特率和gop大小,如果使用udp并体验数据包丢失,请尝试使用tcp。实际上,对于Wireshark,您应该能够看到RTP序列号是否是顺序的。关于参考帧的错误(可能由丢弃/重新排序帧或错误编码引起)是指示图像损坏和可见伪像的错误。 – 2015-02-25 09:24:57

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几个随机的东西可以尝试:如果你从不同的gst-launch实例启动它们,管道行为是否有不同?您是否尝试将“sync = false”添加到autovideosink?你有没有试过增加rtspsrc上的延迟参数?您是否可以访问GStreamer的新版本?您使用的编码器设置是什么? – mpr 2015-02-25 13:54:26

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Rudolfs,它使用UDP。没有丢包,每个包都按顺序接收。我不知道如何改变配置文件,他们是廉价的IP摄像机,只有Windows可用的基本软件。至少我可以用这个软件改变摄像头的IP地址。我正在使用L4T(Linux for Tegra,Ubuntu 14.04)。 – masana 2015-02-26 09:27:11

回答

3

经过几天的深度搜索和一些尝试,我直接打开使用gstreamer-0.10 API。首先,我学会了如何将它与http://docs.gstreamer.com/pages/viewpage.action?pageId=327735

的教程一起使用。对于大多数教程,您只需要安装libgstreamer0.10-dev和其他一些软件包。我安装了所有的人:

sudo apt-get install libgstreamer0* 

然后复制您想尝试进入从那里.c文件的位置(在一些例子中,你必须在文件夹中的终端.c文件和类型的例子的代码添加更多库到pkg-config):

gcc basic-tutorial-1.c $(pkg-config --cflags --libs gstreamer-0.10) -o basic-tutorial-1.c 

之后,我不觉得迷路了,我开始尝试混合一些c和C++代码。您可以使用适当的g ++命令或使用CMakeLists.txt或您想要的方式编译它...在使用nVidia Jetson TK1进行开发时,我使用Nsight Eclipse Edition,并且需要将项目属性正确配置为能够使用gstreamer-0.10 libs和openCV库。

混合一些代码,最后我能够实时捕获我的两个IP摄像机的流而没有明显的延迟,没有任何帧中的错误解码,并且两个流同步。唯一剩下的是我还没有解决的问题是在颜色帧的获取,而不是在灰度时(我已经用“分段故障”结果的其它CV_值试过):

v = Mat(Size(640, 360),CV_8U, (char*)GST_BUFFER_DATA(gstImageBuffer)); 

的完整源代码接下来我使用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。您可以同时复制用于捕获多台摄像机的对象和方法。

#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:[email protected]: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; 
} 

我希望这有助于! ;)