Streaming media playback using gst-launch in the Gstreamer framework

Streaming media playback using gst-launch in the Gstreamer framework

Gstreamer is a set of open source streaming media framework, which can also be used for streaming media development. Gstreamer is written based on the glib library. It needs to pack multiple elements with different functions into a cabinet (bin). Before writing the program, we can use a small tool gst-launch provided by Gstreamer to verify our ideas in the terminal to verify whether the components we selected and the configured pipeline are correct. If the verification is correct, the coding is a small case for all of you. La!

Without further ado, let's go directly to our topic, and list some examples of my previous learning to use Gstreamer for streaming media development, all implemented using gst-launch.

1. RDP protocol realizes mp3 playback

1. Server

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. Receiver

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

2. Camera remote monitoring

1. The sender

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. Receiver

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

3. Audio streaming media transmission (

1. The sender

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. Receiver

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 \

teaaudio. ! queue2 ! ffdec_aac ! audioconvert ! caraudiosink

4. Video files only transmit audio streams

1. The sender

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. Receiver

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

Five, audio and video mixed streaming media

1. The sender

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. Receiver

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 \

teaaudio. ! 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

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