Gstreamer框架中使用gst-launch進行流媒體播放

Gstreamer是一套開源的流媒體框架，用其也可以進行流媒體開發，Gstreamer是基於glib庫編寫的，需要將多個不同功能的元件(element)裝進一個箱櫃(bin)中，在進行程式編寫前，我們可以使用Gstreamer提供的一個小工具gst-launch在終端中在驗證我們的想法，驗證我們選擇的元件和配置的管道是否正確，如果驗證正確，編碼對於各位小夥伴們就是小case啦！

不用廢話了，直接進入我們的主題，一下列出了我以前學習使用Gstreamer進行流媒體開發過程中的一些例項，都是使用gst-launch實現的。

一、RDP協議實現mp3播放

1、服務端

gst-launch filesrc location=/mnt/hgfs/fedshare/source/audio/test.mp3 ! mad! audioconvert \

! audio/x-raw-int,channels=1,depth=16,width=16, rate=44100 !rtpL16pay ! udpsink host=127.0.0.1 port=5000

2、接收端

gst-launch udpsrc port=5000 ! "application/x-rtp,media=(string)audio,clock-rate=(int)44100, width=16, height=16, \

encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1,channel-positions=(int)1, payload=(int)96" ! gstrtpjitterbufferdo-lost=true ! rtpL16depay ! audioconvert ! alsasink sync=false

二、攝像頭遠端監控

1、傳送端

gst-launch -v gstrtpbin name=rtpbin v4l2src device=/dev/video0 ! videorate! videoscale ! ffmpegcolorspace ! 'video/x-raw-yuv, width=(int)320,height=(int)240, framerate=(fraction)15/1' ! \ rtpvrawpay !rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 ! multiudpsinkclients="127.0.0.1:9996" rtpbin.send_rtcp_src_0 ! multiudpsinkclients="127.0.0.1:9997" sync=false async=false udpsrc port=10000 !rtpbin.recv_rtcp_sink_0

2、接收端

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

三、音訊流媒體傳輸(

1、傳送端

gst-launch -v gstrtpbin name=rtpbin latency=100 \

uridecodebin uri=file:///mnt/hgfs/fedshare/source/audio/test.mp3name=decoder \

decoder. ! tee name=tee0 \

tee0. ! queue2 ! audioconvert ! ffenc_aac ! rtpmp4apay pt=96 ! queue2 !rtpbin.send_rtp_sink_1 rtpbin.send_rtp_src_1 ! queue2 ! udpsink host=127.0.0.1port=5002 async=false

2、接收端

gst-launch -v gstrtpbin name=rtpbin latency=100 \

udpsrccaps="application/x-rtp,media=(string)audio,clock-rate=(int)44100,encoding-name=(string)MP4A-LATM,cpresent=(string)0,config=(string)40002420,ssrc=(uint)4204015526,payload=(int)96,clock-base=(uint)4274968681,seqnum-base=(uint)44368"\

port=5002 ! queue2 ! rtpbin.recv_rtp_sink_1 rtpbin. ! rtpmp4adepay ! teename=teeaudio \

teeaudio. ! queue2 ! ffdec_aac ! audioconvert ! autoaudiosink

四、視訊檔案只傳輸音訊流

1、傳送端

gst-launch -v gstrtpbin name=rtpbin latency=100 \

uridecodebin uri=file:///mnt/hgfs/fedshare/source/video/loveStyle.mp4name=decoder \

decoder. ! tee name=tee0 \

tee0. ! queue2 ! ffmpegcolorspace ! x264enc ! rtph264pay pt=96 ! queue2 !rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 ! queue2 ! udpsink host=127.0.0.1port=5000 async=false \

rtpbin.send_rtcp_src_0 ! queue2 ! udpsink host=127.0.0.1 port=5001async=false \

udpsrc port=5005 ! queue2 ! rtpbin.recv_rtcp_sink_0

2、接收端

gst-launch -v gstrtpbin name=rtpbin latency=100 \

udpsrc caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264,ssrc=(uint)340005641,payload=(int)96,clock-base=(uint)2625017333,seqnum-base=(uint)5894"port=5000 \

! queue ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtph264depay ! teename=teevideo \

teevideo. ! queue ! ffdec_h264 ! videoscale ! ffmpegcolorspace !xvimagesink \

udpsrc port=5001 ! queue ! rtpbin.recv_rtcp_sink_0 rtpbin.send_rtcp_src_0! queue ! udpsink host=127.0.0.1 port=5005 sync=true async=false

五、音視訊混合流媒體

1、傳送端

gst-launch -v gstrtpbin name=rtpbin latency=100 \

uridecodebin uri=file:///mnt/hgfs/fedshare/source/video/loveStyle.mp4name=decoder \

decoder. ! tee name=tee0 \

tee0. ! queue2 ! ffmpegcolorspace ! x264enc ! rtph264pay pt=96 ! queue2 !rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 ! queue2 ! udpsink host=127.0.0.1port=5000 async=false \

rtpbin.send_rtcp_src_0 ! queue2 ! udpsink host=127.0.0.1 port=5001async=false \

udpsrc port=5005 ! queue2 ! rtpbin.recv_rtcp_sink_0 \

decoder. ! tee name=tee1 \

tee1. ! queue2 ! audioconvert ! ffenc_aac ! rtpmp4apay pt=96 ! queue2 !rtpbin.send_rtp_sink_1 rtpbin.send_rtp_src_1 ! queue2 ! udpsink host=127.0.0.1port=5002 async=false \

rtpbin.send_rtcp_src_1 ! queue2 ! udpsink host=127.0.0.1 port=5003async=false \

udpsrc port=5006 ! queue2 ! rtpbin.recv_rtcp_sink_1

2、接收端

gst-launch -v gstrtpbin name=rtpbin latency=100 \

udpsrccaps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264,ssrc=(uint)340005641,payload=(int)96,clock-base=(uint)2625017333,seqnum-base=(uint)5894"port=5000 \

! queue2 ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtph264depay ! teename=teevideo \

teevideo. ! queue2 ! ffdec_h264 ! videoscale ! ffmpegcolorspace ! xvimagesink\

udpsrc port=5001 ! queue2 ! rtpbin.recv_rtcp_sink_0 rtpbin.send_rtcp_src_0! queue2 ! udpsink host=127.0.0.1 port=5005 sync=true async=false \

udpsrccaps="application/x-rtp,media=(string)audio,clock-rate=(int)44100,encoding-name=(string)MP4A-LATM,cpresent=(string)0,config=(string)40002420,ssrc=(uint)4204015526,payload=(int)96,clock-base=(uint)4274968681,seqnum-base=(uint)44386"port=5002 ! queue2 ! rtpbin.recv_rtp_sink_1 rtpbin. ! rtpmp4adepay ! teename=teeaudio \

teeaudio. ! queue2 ! ffdec_aac ! audioconvert ! autoaudiosink \

udpsrc port=5003 ! queue2 ! rtpbin.recv_rtcp_sink_1 rtpbin.send_rtcp_src_1! queue2 ! udpsink host=127.0.0.1 port=5006 sync=true async=false