I use the gstreamer to decode the H264, when I use the pipeline like this:
gst-launch-1.0 udpsrc uri=udp://0.0.0.0:15550 caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,payload=(int)33,encoding-name=(string)MP2T" ! .recv_rtp_sink_0 rtpbin latency=800 ! rtpmp2tdepay ! tsdemux name=demux demux. ! h264parse ! queue ! omxh264dec ! vspfilter ! video/x-raw,width=800,height=480 ! waylandsink sync=false max-lateness=-1 demux. ! aacparse ! queue max-size-buffers=8192000 max-size-time=2000000000 ! faad ! alsasink device=media
there would be only about 200ms delay.
And when I set the sync=true, like this:
gst-launch-1.0 udpsrc uri=udp://0.0.0.0:15550 caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,payload=(int)33,encoding-name=(string)MP2T" ! .recv_rtp_sink_0 rtpbin latency=800 ! rtpmp2tdepay ! tsdemux name=demux demux. ! h264parse ! queue ! omxh264dec ! vspfilter ! video/x-raw,width=800,height=480 ! waylandsink sync=true max-lateness=-1 demux. ! aacparse ! queue max-size-buffers=8192000 max-size-time=2000000000 ! faad ! alsasink device=media
the dalay would reach 1200ms
I have no idea about it.
Related
I'm using gstreamer to make a picture-in-picture composition of two rtmp inputs into an rtmp output.
I've managed to create a pipeline that works very well when both streams are offline
However, when one of the rtmp streams is not live when starting the pipeline - the pipeline does not start.
Does anyone know how to overcome this issue, and make sure the pipeline is not blocked if one rtmp source is offline?
You may have to insert identity into compositor input sub-pipelines.
Simulating 2 sources on localhost with:
gst-launch-1.0 videotestsrc ! x264enc insert-vui=1 ! h264parse config-interval=1 ! mpegtsmux ! rtpmp2tpay ! udpsink port=5004
(and second source the same to port 5005), the following runs fine for 0, 1 or 2 sources active at launch time:
gst-launch-1.0 -v \
udpsrc port=5004 ! application/x-rtp,media=video,encoding-name=MP2T,clock-rate=90000,payload=33 ! rtpjitterbuffer latency=300 ! rtpmp2tdepay ! tsdemux ! h264parse ! avdec_h264 ! videoconvert ! identity ! queue ! comp.sink_0 \
udpsrc port=5005 ! application/x-rtp,media=video,encoding-name=MP2T,clock-rate=90000,payload=33 ! rtpjitterbuffer latency=300 ! rtpmp2tdepay ! tsdemux ! h264parse ! avdec_h264 ! videoconvert ! identity ! queue ! comp.sink_1 \
compositor name=comp sink_0::xpos=0 sink_0::ypos=0 sink_0::width=320 sink_0::height=240 sink_1::xpos=0 sink_1::ypos=240 sink_1::width=320 sink_1::height=240 ! video/x-raw,width=320,height=480 ! videoconvert ! xvimagesink
For rtmp, with mpeg audio from first source, it would be something like:
gst-launch-1.0 -v \
rtmpsrc <your source1 and options> ! flvdemux name=demux0 ! queue ! h264parse ! avdec_h264 ! videoconvert ! identity ! queue ! comp.sink_0 \
rtmpsrc <your source2 and options> ! flvdemux name=demux1 ! queue ! h264parse ! avdec_h264 ! videoconvert ! identity ! queue ! comp.sink_1 \
compositor name=comp sink_0::xpos=0 sink_0::ypos=0 sink_0::width=320 sink_0::height=240 sink_1::xpos=0 sink_1::ypos=240 sink_1::width=320 sink_1::height=240 ! video/x-raw,width=320,height=480 ! videoconvert ! autovideosink \
demux0. ! queue ! audio/mpeg ! decodebin ! audioconvert ! audioresample ! autoaudiosink
I cant save audio from stream I get only video in file. I suspect that I do not need two filesink in pipeline or there is some problem two different mux.
I tried to use autoadiosink and autovideosink and they works successfully.
autoadiosink and autovideosink pipeline:
gst-launch-1.0 rtspsrc location=rtsp://184.72.239.149/vod/mp4:BigBuckBunny_115k.mov latency=0 droponlatency=1 name=rtp_source ! queue ! rtph264depay ! decodebin ! videoconvert ! autovideosink rtp_source. ! queue ! decodebin ! autoaudiosink
Save to file filesink pipeline:
gst-launch-1.0 rtspsrc location=rtsp://184.72.239.149/vod/mp4:BigBuckBunny_115k.mov latency=0 droponlatency=1 name=rtp_source ! queue ! rtph264depay ! decodebin ! vp8enc ! webmmux ! filesink location=BigBuckBunny_115k.webm rtp_source. ! "application/x-rtp, media=(string)audio" ! queue ! decodebin ! vorbisenc ! oggmux ! filesink location=BigBuckBunny_115k.webm
I want to get also audio in resulting file.
You just reuse the existing mux - so that the vorbis is put into the webmmux too:
gst-launch-1.0 rtspsrc location=rtsp://184.72.239.149/vod/mp4:BigBuckBunny_115k.mov \
latency=0 droponlatency=1 name=rtp_source ! queue ! rtph264depay ! decodebin ! \
vp8enc ! webmmux name=mux ! filesink location=BigBuckBunny_115k.webm rtp_source. ! \
application/x-rtp, media=(string)audio" ! queue ! decodebin ! vorbisenc ! mux.
I'm trying to record audio and video from internal webcam and mic to segmented files with gstreamer.
It works to a single file by doing:
gst-launch-1.0 -e avfvideosrc !
video/x-raw ! vtenc_h264 ! h264parse ! queue !
mpegtsmux name=mux ! filesink location=test.mp4 osxaudiosrc !
decodebin ! audioconvert ! faac ! aacparse ! queue ! mux.
It doesn't work when doing:
gst-launch-1.0 -e avfvideosrc !
video/x-raw ! vtenc_h264 ! h264parse ! queue !
splitmuxsink
muxer=mpegtsmux
location=test%04d.mp4
max-size-time=1000000000
name=mux osxaudiosrc !
decodebin ! audioconvert ! faac ! aacparse ! queue ! mux.
saying erroneous pipeline: could not link queue1 to mux
I'm using gstreamer 1.12.3 on Mac OSX Sierra
Note: The H264/AAC encoding isn't necessary for what I want to achieve, so if there are solutions that only work with e.g. avimux, for whatever reason, that's fine.
EDIT: I've tried this on a windows machine with the same error.
gst-launch-1.0 -ev ksvideosrc ! video/x-raw !
videoconvert ! queue !
splitmuxsink max-size-time=1000000000 muxer=avimux name=mux
location=video%04d.avi autoaudiosrc !
decodebin ! audioconvert ! queue ! mux.
Just like on Mac, replacing splitmuxsink with avimux ! filesink works. I'm sure I'm just missing out on some 'pipeline' logic so any clarifiction that can push me in the right direction would be helpful.
I needed to send the audio stream to the audio track of the muxer like so: mux.audio_0
gst-launch-1.0 -ev ksvideosrc ! video/x-raw !
videoconvert ! queue !
splitmuxsink max-size-time=1000000000 muxer=avimux name=mux
location=video%04d.avi autoaudiosrc !
decodebin ! audioconvert ! queue ! mux.audio_O
This happens when the documentation should be clear but you're missing out on some basic knowledge on how to interpret it.
I am trying to construct a RTSP pipeline on the client side to receive audio and video streams on android platform
Only video pipeline works fine
data->pipeline = gst_parse_launch("rtspsrc location=rtsp://192.168.1.100:8554/ss ! gstrtpjitterbuffer ! rtph264depay ! h264parse ! amcviddec-omxtiducati1videodecoder ! ffmpegcolorspace ! autovideosink",&error);
I need to receive audio streams also, so I tried with below pipeline
gst-launch rtspsrc location=rtsp://192.168.1.100:8554/ss demux. ! queue ! rtph264depay ! h264parse ! ffdec_h264 ! autovideosink demux. ! queue ! rtpmp4gdepay ! aacparse ! ffdec_aac ! audioconvert ! autoaudiosink
Gstreamer throws error saying no element "demux"
Please let me know proper rtsp pipeline to receive audio and video streams on android
Please try this, (tested):
gst-launch rtspsrc location=rtsp://192.168.1.100:8554/ss name=demux. ! queue ! rtph264depay ! h264parse ! ffdec_h264 ! autovideosink demux. ! queue ! rtpmp4gdepay ! aacparse ! ffdec_aac ! audioconvert ! autoaudiosink
I want to transcode and resize mp4.(mp4-h264_1920x1080/aac => mp4-h264_640x480/mp3) using gstreamer. I wrote down this command.
$ gst-launch-0.10 filesrc location=./gain_1.mp4 ! qtdemux name=demux demux.video_00 ! queue ! ffdec_h264 ! videoscale ! 'video/x-raw-yuv,width=640,height=480' ! x264enc ! queue ! qtmux name=mux mux.video_0 demux.audio_00 ! queue ! ffdec_aac ! lame bitrate=128 ! queue ! mux.audio_0 mux. ! filesink location=0000.mp4 –v -e
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
Redistribute latency...
^CCaught interrupt -- handling interrupt.
Interrupt: Stopping pipeline ...
(gst-launch-0.10:17958): GLib-CRITICAL **: Source ID 1 was not found when attempting to remove it
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
Freeing pipeline ...
which didn't work.
Transcoding video-only works :
gst-launch-0.10 filesrc location=./gain_1.mp4 ! qtdemux name=demux demux.video_00 ! queue ! ffdec_h264 ! videoscale ! 'video/x-raw-yuv, width=640, height=480' ! x264enc ! queue ! mux. mp4mux name=mux ! filesink location=0000.mp4 –v -e
And transcoding audio-only too:
gst-launch-0.10 filesrc location=./gain_1.mp4 ! qtdemux name=demux demux.audio_00 ! ffdec_aac ! lame bitrate=128 ! queue ! mux. mp4mux name=mux ! filesink location=0000.mp4 –v -e
How can I transcode audio and video with the same command?
#Lionel.J I would like to suggest two improvements:
if possible, use gstreamer-1
your solution reads the source file twice. That's not necessary. Furthermore, the audio and video streams are not synchronized when you do this. You can read both audio and video streams out of qtdemux.
This is a pipeline which does the job with gstreamer-1 and reads the source only once:
gst-launch-1.0 -e filesrc location=/path/to/big_buck_bunny_720p_h264.mov ! \
decodebin name=decode ! \
videoscale ! 'video/x-raw,width=640,height=480' ! \
x264enc ! queue ! mp4mux name=mp4mux ! filesink location=0000.mp4 \
decode. ! audioconvert ! lamemp3enc bitrate=128 ! queue ! mp4mux.
Oh~
I solved this problem.
Next command did good work.
gst-launch-0.10 ffmux_mp4 name=mux ! \
filesink location=0000.mp4 \
filesrc location=./gain_1.mp4 ! qtdemux name=vdemux vdemux.video_00 ! queue ! ffdec_h264 ! videoscale ! 'video/x-raw-yuv, width=640, height=480' ! x264enc ! queue ! mux. \
filesrc location=./gain_1.mp4 ! qtdemux name=ademux ademux.audio_00 ! ffdec_aac ! lame bitrate=128 ! queue ! mux.`