Using gstreamer 1.16.2
I have some requirements (due to older reception equipment) to use audio
with AAC encoding (faac), mepgversion=2, stream-format=adts, but it appears
that mpegtsmux does not allow for that, how else might I go about making a
MPEG-TS stream that I can use with this older equipment?
It's a beast of a pipeline on the command line:
gst-launch-1.0 -vvve \
v4l2src device=/dev/video0 io-mode=2 do-timestamp=true !
"video/x-raw, width=1280, height=720, framerate=60/1, format=UYVY" \
! watchdog timeout=5000 \
! tee name=t1 \
t1. ! queue name=t1_nvvidconv0_queue \
! nvvidconv ! "video/x-raw(memory:NVMM), width=1280,
height=720, framerate=60/1, pixel-aspect-ratio=1/1, format=I420" \
! tee name=t2 \
t2. ! queue name=t2_nvoverlaysink0_queue \
! nvoverlaysink name=main_vid display-id=1 overlay=1
overlay-depth=0 enable-last-sample=false \
t2. ! queue name=t2_omxh264enc0_queue \
! omxh264enc bitrate=6500000 profile=main control-rate=2
EnableStringentBitrate=true iframeinterval=15 vbv-size=0 ! "video/x-h264,
level=(string)4.2, stream-format=(string)byte-stream" \
! h264parse \
! queue name=omxh264enc0_mpegtsmux0_queue \
! mpegtsmux name=tsmux \
! rtpmp2tpay \
! queue name=udpsink_queue \
! udpsink port=11000 async=false sync=false host=127.0.0.1
\
alsasrc device=hw:tegrasndt186ref,1 provide-clock=false !
audio/x-raw,channels=4,rate=48000,format=S32LE \
! audioconvert \
! audioresample \
! volume volume=1.0 \
! queue \
! tee name=t3 \
t3. ! queue \
! audioconvert \
! audioresample !
audio/x-raw,channels=4,rate=96000,format=S16LE \
! volume volume=1.0 \
! faac bitrate=96000 tolerance=500000000 \
! aacparse !
audio/mpeg,channels=4,mpegversion=4,base-profile=lc,stream-format=adts\
! queue ! tsmux.
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