的Gstreamer音頻和視頻RTSP應用

我試圖開發流水線上的應用：的Gstreamer音頻和視頻RTSP應用

GST推出-1.0 rtspsrc位置= 「RTSP：//192.168.3.30：8554/rajvi」延遲= 0名稱= demux解複用器。！隊列！ rtpmp4gdepay！ aacparse！ avdec_aac！ audioconvert！ audioresample！ autoaudiosink demux。！隊列！ rtph264depay！ h264parse！ omxh264dec！ videoconvert！ videoscale！ video/x-raw，width = 176，height = 144！ ximagesink

以下是我所實現的代碼：的#include

靜態無效onPadAdded（GstElement *元件，GstPad *墊，gpointer數據） { gchar *名稱;

name = gst_pad_get_name(pad); 
    g_print("A new pad %s was created\n", name); 
    GstCaps * p_caps = gst_pad_get_pad_template_caps (pad); 

    gchar * description = gst_caps_to_string(p_caps); 
    g_free(description); 

    GstElement *depay = GST_ELEMENT(data); 
    if(gst_element_link_pads(element, name, depay, "sink") == 0) 
    { 
      g_print("cb_new_rtspsrc_pad : failed to link elements \n"); 
    } 

    g_free(name); 
} 

int main(int argc, char *argv[]) { 
    GstElement *source, *videosink, *audio, *video, *convert, *pipeline, *audioDepay, *audioQueue, *videoQueue, 
       *audioParse, *audioDecode, *audioConvert, *audioResample, *audioSink, *videoDepay, *videoParser, *videoDecode, *videoConvert, *videoScale, *videoSink; 
    GstCaps *capsFilter; 
    GstBus *bus; 
    GstMessage *msg; 
    GstPad *pad; 
    GstPad *sinkpad,*ghost_sinkpad; 
    gboolean link_ok; 
    GstStateChangeReturn ret; 

    /* Initialize GStreamer */ 
    gst_init (&argc, &argv); 


    /* Create Elements */ 
    pipeline = gst_pipeline_new("rtsp-pipeline"); 
    source = gst_element_factory_make ("rtspsrc", "source"); 


    /*audio bin*/ 
    audioQueue = gst_element_factory_make ("queue", "audio-queue"); 
    audioDepay = gst_element_factory_make ("rtpmp4gdepay", "audio-depayer"); 
    audioParse = gst_element_factory_make ("aacparse", "audio-parser"); 
    audioDecode = gst_element_factory_make ("avdec_aac", "audio-decoder"); 
    audioConvert = gst_element_factory_make ("audioconvert", "aconv"); 
    audioResample = gst_element_factory_make ("audioresample", "audio-resample"); 
    audioSink = gst_element_factory_make ("autoaudiosink", "audiosink"); 

    if (!audioQueue || !audioDepay || !audioParse || !audioConvert ||  !audioResample || !audioSink) 
    { 
      g_printerr("Cannot create audio elements \n"); 
      return 0; 
g_object_set(source, "location", "rtsp://192.168.3.30:8554/rajvi", NULL); 
    g_object_set(source, "latency", 0, NULL); 

    g_signal_connect(G_OBJECT(source), "pad-added", G_CALLBACK(onPadAdded), audioDepay); 

    gst_bin_add_many(GST_BIN(pipeline), source, audioQueue, audioDepay, audioParse, audioDecode, 
        audioConvert, audioResample, audioSink, NULL); 

    if (!gst_element_link_many(audioQueue, audioDepay, audioParse, audioDecode, audioConvert, audioResample, audioSink, NULL)) 
    { 
      g_printerr("Error linking fields ...1 \n"); 
      return 0; 
    } 

    video = gst_bin_new ("videobin"); 
    videoQueue = gst_element_factory_make ("queue", "video-queue"); 
    videoDepay= gst_element_factory_make ("rtph264depay", "video-depayer"); 
    videoParser = gst_element_factory_make ("h264parse", "video-parser"); 
    videoDecode = gst_element_factory_make ("omxh264dec", "video-decoder"); 
    videoConvert = gst_element_factory_make("videoconvert", "convert"); 
    videoScale = gst_element_factory_make("videoscale", "video-scale"); 
    videoSink = gst_element_factory_make("ximagesink", "video-sink"); 
    capsFilter = gst_caps_new_simple("video/x-raw", 
        "width", G_TYPE_INT, 176, 
        "height", G_TYPE_INT, 144, 
        NULL); 

    if (!videoQueue || !videoDepay || !videoParser || !videoDecode || !videoConvert || !videoScale || !videoSink || !capsFilter) 
    { 
      g_printerr("Cannot create video elements \n"); 
      return 0; 
    } 

    gst_bin_add_many(GST_BIN(video),videoQueue, videoDepay, videoParser, videoDecode, videoConvert, videoScale, 
        videosink, NULL); 
    /* set property value */ 
    link_ok = gst_element_link_filtered(videoConvert,videosink, capsFilter); 
    gst_caps_unref (capsFilter); 
    if (!link_ok) { 
      g_warning ("Failed to link element1 and element2!"); 
    } 

    sinkpad = gst_element_get_static_pad (videoConvert, "sink"); 
    ghost_sinkpad = gst_ghost_pad_new ("sink", sinkpad); 
    gst_pad_set_active (ghost_sinkpad, TRUE); 
    gst_element_add_pad (video, ghost_sinkpad); 

    if (!gst_element_link_many(videoQueue, videoDepay, videoParser, videoDecode, videoScale, NULL)) 
    { 
      g_printerr("Error linking fields... 2 \n"); 
      return 0; 
    } 

    gst_bin_add_many (GST_BIN(pipeline), video,NULL); 
    /* Start playing */ 
    gst_element_set_state (pipeline, GST_STATE_PLAYING); 

    /* Wait until error or EOS */ 
    bus = gst_element_get_bus (pipeline); 
    msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE, GST_MESSAGE_ERROR | GST_MESSAGE_EOS); 

    /* Free resources */ 
    if (msg != NULL) 
      gst_message_unref (msg); 
    gst_object_unref (bus); 
    gst_element_set_state (pipeline, GST_STATE_NULL); 
    gst_object_unref (pipeline); 
    return 0; 
}

得到錯誤鏈接pipeline->音頻>視頻箱

來源

2017-01-16 RajviK

如果你把在管道斌視頻和音頻都在一起，那麼你可以做到這一點。找出你的視頻和音頻上限，並且應該能夠鏈接它們。

// ---------------------------------- 
// pad-added signal 
// ---------------------------------- 

    static void onPadAdded(GstElement* element, GstPad* pad, gpointer user_data) 
    { 
     gchar *name; 
     GstCaps * p_caps; 
     GstElement* nextElement; 
     GstElement* pipeline = (GstElement*)user_data; 
     name = gst_pad_get_name(pad); 
     g_print("A new pad %s was created\n", name); 
     p_caps = gst_pad_get_pad_template_caps(pad); 

     if (strstr(name, "[CAPS FOR VIDEO CONTAIN]") != NULL) 
     { 
      std::cout << std::endl << "------------------------ Video -------------------------------" << std::endl; 

     nextElement = gst_bin_get_by_name(GST_BIN(pipeline), "video-depayer"); 
     } 

     else if (strstr(name, "[CAPS FOR AUDIO CONTAIN]") != NULL) 
     { 
     std::cout << std::endl << "------------------------ Audio -------------------------------" << std::endl; 

     nextElement = gst_bin_get_by_name(GST_BIN(pipeline), "audio-depayer"); 

     } 
     if (nextElement != NULL) 
     { 
     if (!gst_element_link_filtered(element, nextElement, p_caps)) 
      //if (!gst_element_link_pads_filtered(element, name, nextElement, "sink", p_caps)) 
     { 
      std::cout << std::endl << "Failed to link video element to src to sink" << std::endl; 
     } 
     gst_object_unref(nextElement); 
    } 

    g_free(name); 
    gst_caps_unref(p_caps); 
    } 
// ---------------------------------- 
// main 
// ---------------------------------- 

    int main(int argc, char *argv[]) 
    { 
    GstElement *source, *videosink, *audio,*convert, *pipeline, *audioDepay, *audioQueue, *videoQueue, 
     *audioParse, *audioDecode, *audioConvert, *audioResample, *audioSink, *videoDepay, *videoParser, *videoDecode, *videoConvert, *videoScale, *videoSink; 
    GstCaps *capsFilter; 
    GstBus *bus; 
    GstMessage *msg; 
    GstPad *pad; 
    gboolean link_ok; 
    GstStateChangeReturn ret; 

    /* Initialize GStreamer */ 
    gst_init(&argc, &argv); 


    /* Create Elements */ 
    pipeline = gst_pipeline_new("rtsp-pipeline"); 
    source = gst_element_factory_make("rtspsrc", "source"); 


    /*audio bin*/ 
    audioQueue = gst_element_factory_make("queue", "audio-queue"); 
    audioDepay = gst_element_factory_make("rtpmp4gdepay", "audio-depayer"); 
    audioParse = gst_element_factory_make("aacparse", "audio-parser"); 
    audioDecode = gst_element_factory_make("avdec_aac", "audio-decoder"); 
    audioConvert = gst_element_factory_make("audioconvert", "aconv"); 
    audioResample = gst_element_factory_make("audioresample", "audio-resample"); 
    audioSink = gst_element_factory_make("autoaudiosink", "audiosink"); 

    if (!audioQueue || !audioDepay || !audioParse || !audioConvert || !audioResample || !audioSink) 
    { 
     g_printerr("Cannot create audio elements \n"); 
     return 0; 
     g_object_set(source, "location", "rtsp://192.168.3.30:8554/rajvi", NULL); 
     g_object_set(source, "latency", 0, NULL); 

     g_signal_connect(G_OBJECT(source), "pad-added", G_CALLBACK(onPadAdded), pipeline); 

     gst_bin_add_many(GST_BIN(pipeline), source, audioQueue, audioDepay, audioParse, audioDecode, 
      audioConvert, audioResample, audioSink, NULL); 

     if (!gst_element_link_many(audioQueue, audioDepay, audioParse, audioDecode, audioConvert, audioResample, audioSink, NULL)) 
     { 
      g_printerr("Error linking fields ...1 \n"); 
      return 0; 
     } 

     videoQueue = gst_element_factory_make("queue", "video-queue"); 
     videoDepay = gst_element_factory_make("rtph264depay", "video-depayer"); 
     videoParser = gst_element_factory_make("h264parse", "video-parser"); 
     videoDecode = gst_element_factory_make("omxh264dec", "video-decoder"); 
     videoConvert = gst_element_factory_make("videoconvert", "convert"); 
     videoScale = gst_element_factory_make("videoscale", "video-scale"); 
     videoSink = gst_element_factory_make("ximagesink", "video-sink"); 
     capsFilter = gst_caps_new_simple("video/x-raw", 
      "width", G_TYPE_INT, 176, 
      "height", G_TYPE_INT, 144, 
      NULL); 

     if (!videoQueue || !videoDepay || !videoParser || !videoDecode || !videoConvert || !videoScale || !videoSink || !capsFilter) 
     { 
      g_printerr("Cannot create video elements \n"); 
      return 0; 
     } 

     gst_bin_add_many(GST_BIN(pipeline), videoQueue, videoDepay, videoParser, videoDecode, videoConvert, videoScale, 
      videosink, NULL); 
     /* set property value */ 
     link_ok = gst_element_link_filtered(videoConvert, videosink, capsFilter); 
     gst_caps_unref(capsFilter); 
     if (!link_ok) { 
      g_warning("Failed to link element1 and element2!"); 
     } 

     if (!gst_element_link_many(videoQueue, videoDepay, videoParser, videoDecode, videoScale, NULL)) 
     { 
      g_printerr("Error linking fields... 2 \n"); 
      return 0; 
     } 

     /* Start playing */ 
     gst_element_set_state(pipeline, GST_STATE_PLAYING); 

     /* Wait until error or EOS */ 
     bus = gst_element_get_bus(pipeline); 
     msg = gst_bus_timed_pop_filtered(bus, GST_CLOCK_TIME_NONE,(GstMessageType)(GST_MESSAGE_ERROR | GST_MESSAGE_EOS)); 

     /* Free resources */ 
     if (msg != NULL) 
      gst_message_unref(msg); 
     gst_object_unref(bus); 
     gst_element_set_state(pipeline, GST_STATE_NULL); 
     gst_object_unref(pipeline); 
     return 0; 
    } 
    }

來源

2017-12-20 18:32:36 user2412974

的Gstreamer音頻和視頻RTSP應用

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