Gstreamer实现摄像头的远程采集，udp传输，本地显示和保存为AVI文件 接收保存显示端

经过两个星期的努力终于完成 Gstreamer实现摄像头的远程采集，udp传输，本地显示和保存为AVI文件，的C语言程序，现在分享给大家，欢迎大家评论指正

由于本程序存在录制时间短但保存成文件的播放长度很长的问题，希望知道的高手们指点一下解决的方法，在此先谢谢了！！！！

recv-display-avifile:

gst-launch  udpsrc caps=" application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)RAW, sampling=(string)YCbCr-4:2:0, depth=(string)8, width=(string)320, height=(string)240, colorimetry=(string)SMPTE240M, ssrc=(guint)4294234526, payload=(int)96, clock-base=(guint)520513122, seqnum-base=(guint)28177" port=9996 ! queue ! rtpvrawdepay  ! queue ! tee name="splitter" ! queue ! ffmpegcolorspace ! autovideosink splitter. ! queue ! ffmpegcolorspace ! jpegenc ! avimux ! filesink location=osug-udp-2.avi

C code:

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1. #include <string.h>  
2. #include <math.h>  
3.   
4. #include <gst/gst.h>  
5.   
6. /* the caps of the sender RTP stream. This is usually negotiated out of band with  
7.  * SDP or RTSP. */  
8. #define VIDEO_CAPS "application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)RAW, sampling=(string)YCbCr-4:2:0, depth=(string)8, width=(string)320, height=(string)240, colorimetry=(string)SMPTE240M"  
9. //#define VIDEO_CAPS "application/x-rtp,media=video,clock-rate=9000,encoding-name=H264"  
10.   
11.   
12. #define AVINAME "camera.avi"   
13. #define PORT 9996  
14. #define VIDEO_SINK  "autovideosink"  
15. /* the destination machine to send RTCP to. This is the address of the sender and  
16.  * is used to send back the RTCP reports of this receiver. If the data is sent  
17.  * from another machine, change this address. */  
18. #define DEST_HOST "127.0.0.1"  
19.   
20. /* print the stats of a source */  
21. static void print_source_stats (GObject * source) {  
22.   GstStructure *stats;  
23.   gchar *str;  
24.   
25.   g_return_if_fail (source != NULL);  
26.   
27.   /* get the source stats */  
28.   g_object_get (source, "stats", &stats, NULL);  
29.   
30.   /* simply dump the stats structure */  
31.   str = gst_structure_to_string (stats);  
32.   g_print ("source stats: %s\n", str);  
33.   
34.   gst_structure_free (stats);  
35.   g_free (str);  
36. }  
37.   
38. /* will be called when gstrtpbin signals on-ssrc-active. It means that an RTCP  
39.  * packet was received from another source. */  
40. static void on_ssrc_active_cb (GstElement * rtpbin, guint sessid, guint ssrc, GstElement * depay) {  
41.   
42.     GObject *session, *isrc, *osrc;  
43.     g_print ("got RTCP from session %u, SSRC %u\n", sessid, ssrc);  
44.   
45.   /* get the right session */  
46.   g_signal_emit_by_name (rtpbin, "get-internal-session", sessid, &session);  
47.   
48.   /* get the internal source (the SSRC allocated to us, the receiver */  
49.   g_object_get (session, "internal-source", &isrc, NULL);  
50.   print_source_stats (isrc);  
51.   
52.   /* get the remote source that sent us RTCP */  
53.   g_signal_emit_by_name (session, "get-source-by-ssrc", ssrc, &osrc);  
54.   print_source_stats (osrc);  
55. }  
56.   
57. /* will be called when rtpbin has validated a payload that we can depayload */  
58. static void  
59. pad_added_cb (GstElement * rtpbin, GstPad * new_pad, GstElement * depay)  
60. {  
61.   GstPad *sinkpad;  
62.   GstPadLinkReturn lres;  
63.   
64.   g_print ("new payload on pad: %s\n", GST_PAD_NAME (new_pad));  
65.   
66.   sinkpad = gst_element_get_static_pad (depay, "sink");  
67.   g_assert (sinkpad);  
68.   
69.   lres = gst_pad_link (new_pad, sinkpad);  
70.   g_assert (lres == GST_PAD_LINK_OK);  
71.   gst_object_unref (sinkpad);  
72.   
73. }  
74.   
75.   
76. int main (int argc, char *argv[])  
77. {  
78.   GstElement *rtpbin, *rtpsrc, *rtcpsrc, *rtcpsink;  
79.   GstElement *videodepay,  
80.              *videodec,  
81.              *videoqueue,  
82.              //*videores,  
83.              *videoconv,  
84.              *videosink,  
85.   
86.          *tee,  
87.          *aviqueue,  
88.              *aviconv,  
89.          *avidenc,  
90.          *avifmux,  
91.          *avifilesink;  
92.   
93.   
94.   GstElement *pipeline;  
95.   GMainLoop *loop;  
96.   GstCaps *caps;  
97.   gboolean res1,res2;  
98.   GstPadLinkReturn lres;  
99.   GstPad *srcpad, *sinkpad;  
100.   
101.   /* always init first */  
102.   gst_init (&argc, &argv);  
103.   
104.   /* the pipeline to hold everything */  
105.   pipeline = gst_pipeline_new (NULL);  
106.   g_assert (pipeline);  
107.   
108.   /* the udp src and source we will use for RTP and RTCP */  
109.   rtpsrc = gst_element_factory_make ("udpsrc", "rtpsrc");  
110.   g_assert (rtpsrc);  
111.   g_object_set (rtpsrc, "port", PORT, NULL);  
112.   /* we need to set caps on the udpsrc for the RTP data */  
113.   caps = gst_caps_from_string (VIDEO_CAPS);  
114.   g_object_set (rtpsrc, "caps", caps, NULL);  
115.   gst_caps_unref (caps);  
116.   
117.   rtcpsrc = gst_element_factory_make ("udpsrc", "rtcpsrc");  
118.   g_assert (rtcpsrc);  
119.   g_object_set (rtcpsrc, "port", 9997, NULL);  
120.   
121.   rtcpsink = gst_element_factory_make ("udpsink", "rtcpsink");  
122.   g_assert (rtcpsink);  
123.   g_object_set (rtcpsink, "port", 9999, "host", DEST_HOST, NULL);  
124.   /* no need for synchronisation or preroll on the RTCP sink */  
125.   g_object_set (rtcpsink, "async", FALSE, "sync", FALSE, NULL);  
126.   
127.   gst_bin_add_many (GST_BIN (pipeline), rtpsrc, rtcpsrc, rtcpsink, NULL);  
128.   
129.   /* the depayloading and decoding */  
130.   videodepay = gst_element_factory_make ("rtpvrawdepay", "videodepay");  
131.   g_assert (videodepay);  
132.   videoqueue=gst_element_factory_make ("queue","videoqueue");  
133.   g_assert(videoqueue);  
134.   
135.   tee = gst_element_factory_make ("tee","tee");  
136.   g_assert(tee);  
137.   
138.   aviqueue=gst_element_factory_make ("queue","aviqueue");  
139.   g_assert(aviqueue);  
140.   
141. //  videodec = gst_element_factory_make ("ffmpegcolorspace", "videodec");  
142. //  g_assert (videodec);  
143.   /* the audio playback and format conversion */  
144.   videoconv = gst_element_factory_make ("ffmpegcolorspace", "videoconv");  
145.   g_assert (videoconv);  
146. /*  
147.   audiores = gst_element_factory_make ("audioresample", "audiores");  
148.   g_assert (audiores);  
149. */  
150.   videosink = gst_element_factory_make (VIDEO_SINK, "videosink");  
151.   g_assert (videosink);  
152.   
153.   aviconv = gst_element_factory_make ("ffmpegcolorspace","avicinv");  
154.   g_assert (aviconv);  
155.   avidenc = gst_element_factory_make ("jpegenc","avidenc");  
156.   g_assert (aviconv);  
157.   avifmux = gst_element_factory_make ("avimux","avifmux");  
158.   g_assert (avifmux);  
159.   avifilesink = gst_element_factory_make ("filesink","avifilesink");  
160.   g_assert (avifilesink);  
161.     
162.   
163.   g_object_set(avifilesink,"location",AVINAME,NULL);  
164.   
165.   /* add depayloading and playback to the pipeline and link */  
166. //  gst_bin_add_many (GST_BIN (pipeline), videodepay, videoconv, /*videores,*/videoqueue, videosink, aviconv,avidenc,avifmux,avifilename,NULL);  
167.   gst_bin_add_many (GST_BIN (pipeline), videodepay,tee,videoqueue,videoconv,videosink, aviqueue,aviconv,avidenc,avifmux,avifilesink,NULL);  
168.   
169.   res1 = gst_element_link_many (videodepay, tee,videoqueue,videoconv,videosink, NULL);  
170.   g_assert (res1 == TRUE);  
171.   res2 = gst_element_link_many (tee,aviqueue,aviconv,avidenc,avifmux,avifilesink,NULL);  
172.   g_assert (res2 == TRUE);  
173.   
174.   /* the rtpbin element */  
175.   rtpbin = gst_element_factory_make ("gstrtpbin", "rtpbin");  
176.   g_assert (rtpbin);  
177.   
178.   g_object_set (G_OBJECT (rtpbin),"latency",200,NULL);  
179.   
180.   gst_bin_add (GST_BIN (pipeline), rtpbin);  
181.   
182.   /* now link all to the rtpbin, start by getting an RTP sinkpad for session 0 */  
183.   srcpad = gst_element_get_static_pad (rtpsrc, "src");  
184.   sinkpad = gst_element_get_request_pad (rtpbin, "recv_rtp_sink_0");  
185.   lres = gst_pad_link (srcpad, sinkpad);  
186.   g_assert (lres == GST_PAD_LINK_OK);  
187.   gst_object_unref (srcpad);  
188.   
189.   /* get an RTCP sinkpad in session 0 */  
190.   srcpad = gst_element_get_static_pad (rtcpsrc, "src");  
191.   sinkpad = gst_element_get_request_pad (rtpbin, "recv_rtcp_sink_0");  
192.   lres = gst_pad_link (srcpad, sinkpad);  
193.   g_assert (lres == GST_PAD_LINK_OK);  
194.   gst_object_unref (srcpad);  
195.   gst_object_unref (sinkpad);  
196.   
197.   /* get an RTCP srcpad for sending RTCP back to the sender */  
198.   srcpad = gst_element_get_request_pad (rtpbin, "send_rtcp_src_0");  
199.   sinkpad = gst_element_get_static_pad (rtcpsink, "sink");  
200.   lres = gst_pad_link (srcpad, sinkpad);  
201.   g_assert (lres == GST_PAD_LINK_OK);  
202.   gst_object_unref (sinkpad);  
203.   
204.   /* the RTP pad that we have to connect to the depayloader will be created  
205.    * dynamically so we connect to the pad-added signal, pass the depayloader as  
206.    * user_data so that we can link to it. */  
207.   g_signal_connect (rtpbin, "pad-added", G_CALLBACK (pad_added_cb), videodepay);  
208.   
209.   /* give some stats when we receive RTCP */  
210.   //g_signal_connect (rtpbin, "on-ssrc-active", G_CALLBACK (on_ssrc_active_cb),videodepay);  
211.   
212.   /* set the pipeline to playing */  
213.   g_print ("starting receiver pipeline\n");  
214.   gst_element_set_state (pipeline, GST_STATE_PLAYING);  
215.   
216.   /* we need to run a GLib main loop to get the messages */  
217.   loop = g_main_loop_new (NULL, FALSE);  
218.   g_main_loop_run (loop);  
219.   
220.   g_print ("stopping receiver pipeline\n");  
221.   gst_element_set_state (pipeline, GST_STATE_NULL);  
222.   
223.   gst_object_unref (pipeline);  
224.   
225.   return 0;  
226. }  

#include <string.h>
#include <math.h>

#include <gst/gst.h>

/* the caps of the sender RTP stream. This is usually negotiated out of band with
 * SDP or RTSP. */
#define VIDEO_CAPS "application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)RAW, sampling=(string)YCbCr-4:2:0, depth=(string)8, width=(string)320, height=(string)240, colorimetry=(string)SMPTE240M"
//#define VIDEO_CAPS "application/x-rtp,media=video,clock-rate=9000,encoding-name=H264"


#define AVINAME "camera.avi" 
#define PORT 9996
#define VIDEO_SINK  "autovideosink"
/* the destination machine to send RTCP to. This is the address of the sender and
 * is used to send back the RTCP reports of this receiver. If the data is sent
 * from another machine, change this address. */
#define DEST_HOST "127.0.0.1"

/* print the stats of a source */
static void print_source_stats (GObject * source) {
  GstStructure *stats;
  gchar *str;

  g_return_if_fail (source != NULL);

  /* get the source stats */
  g_object_get (source, "stats", &stats, NULL);

  /* simply dump the stats structure */
  str = gst_structure_to_string (stats);
  g_print ("source stats: %s\n", str);

  gst_structure_free (stats);
  g_free (str);
}

/* will be called when gstrtpbin signals on-ssrc-active. It means that an RTCP
 * packet was received from another source. */
static void on_ssrc_active_cb (GstElement * rtpbin, guint sessid, guint ssrc, GstElement * depay) {

    GObject *session, *isrc, *osrc;
    g_print ("got RTCP from session %u, SSRC %u\n", sessid, ssrc);

  /* get the right session */
  g_signal_emit_by_name (rtpbin, "get-internal-session", sessid, &session);

  /* get the internal source (the SSRC allocated to us, the receiver */
  g_object_get (session, "internal-source", &isrc, NULL);
  print_source_stats (isrc);

  /* get the remote source that sent us RTCP */
  g_signal_emit_by_name (session, "get-source-by-ssrc", ssrc, &osrc);
  print_source_stats (osrc);
}

/* will be called when rtpbin has validated a payload that we can depayload */
static void
pad_added_cb (GstElement * rtpbin, GstPad * new_pad, GstElement * depay)
{
  GstPad *sinkpad;
  GstPadLinkReturn lres;

  g_print ("new payload on pad: %s\n", GST_PAD_NAME (new_pad));

  sinkpad = gst_element_get_static_pad (depay, "sink");
  g_assert (sinkpad);

  lres = gst_pad_link (new_pad, sinkpad);
  g_assert (lres == GST_PAD_LINK_OK);
  gst_object_unref (sinkpad);

}


int main (int argc, char *argv[])
{
  GstElement *rtpbin, *rtpsrc, *rtcpsrc, *rtcpsink;
  GstElement *videodepay,
             *videodec,
             *videoqueue,
             //*videores,
             *videoconv,
             *videosink,

	     *tee,
	     *aviqueue,
             *aviconv,
	     *avidenc,
	     *avifmux,
	     *avifilesink;


  GstElement *pipeline;
  GMainLoop *loop;
  GstCaps *caps;
  gboolean res1,res2;
  GstPadLinkReturn lres;
  GstPad *srcpad, *sinkpad;

  /* always init first */
  gst_init (&argc, &argv);

  /* the pipeline to hold everything */
  pipeline = gst_pipeline_new (NULL);
  g_assert (pipeline);

  /* the udp src and source we will use for RTP and RTCP */
  rtpsrc = gst_element_factory_make ("udpsrc", "rtpsrc");
  g_assert (rtpsrc);
  g_object_set (rtpsrc, "port", PORT, NULL);
  /* we need to set caps on the udpsrc for the RTP data */
  caps = gst_caps_from_string (VIDEO_CAPS);
  g_object_set (rtpsrc, "caps", caps, NULL);
  gst_caps_unref (caps);

  rtcpsrc = gst_element_factory_make ("udpsrc", "rtcpsrc");
  g_assert (rtcpsrc);
  g_object_set (rtcpsrc, "port", 9997, NULL);

  rtcpsink = gst_element_factory_make ("udpsink", "rtcpsink");
  g_assert (rtcpsink);
  g_object_set (rtcpsink, "port", 9999, "host", DEST_HOST, NULL);
  /* no need for synchronisation or preroll on the RTCP sink */
  g_object_set (rtcpsink, "async", FALSE, "sync", FALSE, NULL);

  gst_bin_add_many (GST_BIN (pipeline), rtpsrc, rtcpsrc, rtcpsink, NULL);

  /* the depayloading and decoding */
  videodepay = gst_element_factory_make ("rtpvrawdepay", "videodepay");
  g_assert (videodepay);
  videoqueue=gst_element_factory_make ("queue","videoqueue");
  g_assert(videoqueue);

  tee = gst_element_factory_make ("tee","tee");
  g_assert(tee);

  aviqueue=gst_element_factory_make ("queue","aviqueue");
  g_assert(aviqueue);

//  videodec = gst_element_factory_make ("ffmpegcolorspace", "videodec");
//  g_assert (videodec);
  /* the audio playback and format conversion */
  videoconv = gst_element_factory_make ("ffmpegcolorspace", "videoconv");
  g_assert (videoconv);
/*
  audiores = gst_element_factory_make ("audioresample", "audiores");
  g_assert (audiores);
*/
  videosink = gst_element_factory_make (VIDEO_SINK, "videosink");
  g_assert (videosink);

  aviconv = gst_element_factory_make ("ffmpegcolorspace","avicinv");
  g_assert (aviconv);
  avidenc = gst_element_factory_make ("jpegenc","avidenc");
  g_assert (aviconv);
  avifmux = gst_element_factory_make ("avimux","avifmux");
  g_assert (avifmux);
  avifilesink = gst_element_factory_make ("filesink","avifilesink");
  g_assert (avifilesink);
  

  g_object_set(avifilesink,"location",AVINAME,NULL);

  /* add depayloading and playback to the pipeline and link */
//  gst_bin_add_many (GST_BIN (pipeline), videodepay, videoconv, /*videores,*/videoqueue, videosink, aviconv,avidenc,avifmux,avifilename,NULL);
  gst_bin_add_many (GST_BIN (pipeline), videodepay,tee,videoqueue,videoconv,videosink, aviqueue,aviconv,avidenc,avifmux,avifilesink,NULL);

  res1 = gst_element_link_many (videodepay, tee,videoqueue,videoconv,videosink, NULL);
  g_assert (res1 == TRUE);
  res2 = gst_element_link_many (tee,aviqueue,aviconv,avidenc,avifmux,avifilesink,NULL);
  g_assert (res2 == TRUE);

  /* the rtpbin element */
  rtpbin = gst_element_factory_make ("gstrtpbin", "rtpbin");
  g_assert (rtpbin);

  g_object_set (G_OBJECT (rtpbin),"latency",200,NULL);

  gst_bin_add (GST_BIN (pipeline), rtpbin);

  /* now link all to the rtpbin, start by getting an RTP sinkpad for session 0 */
  srcpad = gst_element_get_static_pad (rtpsrc, "src");
  sinkpad = gst_element_get_request_pad (rtpbin, "recv_rtp_sink_0");
  lres = gst_pad_link (srcpad, sinkpad);
  g_assert (lres == GST_PAD_LINK_OK);
  gst_object_unref (srcpad);

  /* get an RTCP sinkpad in session 0 */
  srcpad = gst_element_get_static_pad (rtcpsrc, "src");
  sinkpad = gst_element_get_request_pad (rtpbin, "recv_rtcp_sink_0");
  lres = gst_pad_link (srcpad, sinkpad);
  g_assert (lres == GST_PAD_LINK_OK);
  gst_object_unref (srcpad);
  gst_object_unref (sinkpad);

  /* get an RTCP srcpad for sending RTCP back to the sender */
  srcpad = gst_element_get_request_pad (rtpbin, "send_rtcp_src_0");
  sinkpad = gst_element_get_static_pad (rtcpsink, "sink");
  lres = gst_pad_link (srcpad, sinkpad);
  g_assert (lres == GST_PAD_LINK_OK);
  gst_object_unref (sinkpad);

  /* the RTP pad that we have to connect to the depayloader will be created
   * dynamically so we connect to the pad-added signal, pass the depayloader as
   * user_data so that we can link to it. */
  g_signal_connect (rtpbin, "pad-added", G_CALLBACK (pad_added_cb), videodepay);

  /* give some stats when we receive RTCP */
  //g_signal_connect (rtpbin, "on-ssrc-active", G_CALLBACK (on_ssrc_active_cb),videodepay);

  /* set the pipeline to playing */
  g_print ("starting receiver pipeline\n");
  gst_element_set_state (pipeline, GST_STATE_PLAYING);

  /* we need to run a GLib main loop to get the messages */
  loop = g_main_loop_new (NULL, FALSE);
  g_main_loop_run (loop);

  g_print ("stopping receiver pipeline\n");
  gst_element_set_state (pipeline, GST_STATE_NULL);

  gst_object_unref (pipeline);

  return 0;
}

由于本程序存在录制时间短但保存成文件的播放长度很长的问题，希望知道的高手们指点一下解决的方法，在此先谢谢了！！！！