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:
#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!!!!
程序的发送端在上一篇博客中欢迎浏览!!!!http://blog.csdn.net/zhujinghao09/article/details/8528802