hello folks i have a mkv that want to convert to mp4, ogg and webm
i tried following pipipeline but got stuck
gst-launch-1.0 filesrc location=output.mkv ! matroskademux name=demux demux.video_0 ! jpegparse ! jpegdec ! tee name=vo \
oggmux name=ogg ! filesink location=output.ogg \
qtmux name=mp4 ! filesink location=output.mp4 \
webmmux name=webm ! filesink location=output.webm \
vo. ! queue ! videoconvert ! vp8enc ! progressreport update-freq=1 ! webm. \
vo. ! queue ! videoconvert ! x264enc ! progressreport update-freq=1 ! mp4. \
vo. ! queue ! videoconvert ! theoraenc ! progressreport update-freq=1 ! ogg.
output
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
Redistribute latency...
Redistribute latency...
progressreport2 (00:00:01): 0 seconds
progressreport2 (00:00:02): 0 seconds
progressreport0 (00:00:02): 0 seconds
from what i am seeing there is a problem with the queues, bc its only recognizing 2 and should have 3 of them
moreover if i remove one queue, from the pipeline, seems to be working just fine
ie
gst-launch-1.0 filesrc location=output.mkv ! matroskademux name=demux demux.video_0 ! jpegparse ! jpegdec ! tee name=vo \
oggmux name=ogg ! filesink location=output.ogg \
webmmux name=webm ! filesink location=output.webm \
vo. ! queue ! videoconvert ! vp8enc ! progressreport update-freq=1 ! webm. \
vo. ! queue ! videoconvert ! theoraenc ! progressreport update-freq=1 ! ogg.
output
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
Redistribute latency...
progressreport1 (00:00:01): 0 seconds
progressreport0 (00:00:02): 0 seconds
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
progressreport1 (00:00:02): 0 seconds
progressreport1 (00:00:03): 0 seconds
progressreport1 (00:00:04): 0 seconds
progressreport1 (00:00:05): 0 seconds
progressreport1 (00:00:06): 0 seconds
progressreport0 (00:00:08): 0 seconds
progressreport1 (00:00:09): 0 seconds
please advice
regards
x264enc by default has a latency higher than the default queue sizes causing your pipeline to stall. (It has to consume more data than it currently gets to actually create an output buffer. That way the pipeline will never finish pre-roll).
Two options:
increase buffer size of the queue sizes for all queues.
use x264enc tune=zerolatency for your encoder.
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
How to mix h264 format with audio on webcam with gstreamer?
gst-launch-1.0 -v v4l2src device=/dev/video2 ! video/x-h264,framerate=30/1,width=1920,height=1080 \
! queue ! mux. \
alsasrc device=hw:1 ! queue ! audioconvert ! fdkaacenc \
! mux. matroskamux name=mux ! filesink location=video.mkv
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstAudioSrcClock
ERROR: from element /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: Internal data stream error.
Additional debug info:
../libs/gst/base/gstbasesrc.c(3127): gst_base_src_loop (): /GstPipeline:pipeline0/GstV4l2Src:v4l2src0:
streaming stopped, reason not-negotiated (-4)
Execution ended after 0:00:00.001309727
Setting pipeline to NULL ...
Freeing pipeline ...
Preview works
gst-launch-1.0 -v \
v4l2src device=/dev/video2 ! video/x-h264,framerate=30/1,width=1920,height=1080 ! decodebin ! autovideosink
Audio works
gst-launch-1.0 -v alsasrc device=hw:1 ! queue ! audioconvert ! fdkaacenc ! fdkaacdec ! autoaudiosink
h264parse needed before mux
gst-launch-1.0 -v \
v4l2src device=/dev/video2 ! queue ! video/x-h264,framerate=30/1,width=1920,height=1080 \
! h264parse ! mux. \
alsasrc device=hw:1 ! queue ! audioconvert ! fdkaacenc ! mux. \
matroskamux name=mux ! filesink location=video.mp4
I am trying to reencode the audio part of a MKV file that contains some video/x-h264 and some audio/x-raw. I can't manage to just demux the MKV and remux it. Even simply:
gst-launch-1.0 filesrc location=test.mkv ! matroskademux name=demux \
matroskamux name=mux ! filesink location=test2.mkv \
demux.video_00 ! mux.video_00 \
demux.audio_00 ! mux.audio_00
fails miserably with:
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
WARNING: from element /GstPipeline:pipeline0/GstMatroskaDemux:demux: Delayed linking failed.
Additional debug info:
../gstreamer/gst/parse/grammar.y(506): gst_parse_no_more_pads (): /GstPipeline:pipeline0/GstMatroskaDemux:demux:
failed delayed linking pad video_00 of GstMatroskaDemux named demux to pad video_00 of GstMatroskaMux named mux
WARNING: from element /GstPipeline:pipeline0/GstMatroskaDemux:demux: Delayed linking failed.
Additional debug info:
../gstreamer/gst/parse/grammar.y(506): gst_parse_no_more_pads (): /GstPipeline:pipeline0/GstMatroskaDemux:demux:
failed delayed linking pad audio_00 of GstMatroskaDemux named demux to pad audio_00 of GstMatroskaMux named mux
ERROR: from element /GstPipeline:pipeline0/GstMatroskaDemux:demux: Internal data stream error.
Additional debug info:
../gst-plugins-good/gst/matroska/matroska-demux.c(5715): gst_matroska_demux_loop (): /GstPipeline:pipeline0/GstMatroskaDemux:demux:
streaming stopped, reason not-linked (-1)
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
Freeing pipeline ...
My best attempt at the transcoding mentioned above goes:
gst-launch-1.0 -v filesrc location=test.mkv ! matroskademux name=demux \
matroskamux name=mux ! filesink location=test2.mkv \
demux.video_00 ! queue ! 'video/x-h264' ! h264parse ! mux. \
demux.audio_00 ! queue ! rawaudioparse ! audioconvert ! audioresample ! avenc_aac ! mux.
with the same result. Removing the pad name audio_00 leads to gst being stuck at PREROLLING.
I have seen a few people facing similar problems:
http://gstreamer-devel.966125.n4.nabble.com/Putting-h264-file-inside-a-container-td4668158.html
http://gstreamer-devel.966125.n4.nabble.com/Changing-the-container-format-td3576914.html
As therein, keeping only video or only audio works.
I think the rawaudioparse should not be here. I tried your pipeline and trouble with it too. I just came up with something as I would have done it and it seemed to work:
filesrc location=test.mkv ! matroskademux \
matroskademux0. ! queue ! audioconvert ! avenc_aac ! matroskamux ! filesink location=test2.mkv \
matroskademux0. ! queue ! h264parse ! matroskamux0.
Audio in my case was:
Stream #0:0(eng): Audio: pcm_f32le, 44100 Hz, 2 channels, flt, 2822 kb/s (default)
Another format may require addiitonal transformations..
The problem is that the pads video_00 and audio_00 have been renamed video_0 and audio_0. This can be seen using gst-inspect-1.0 matroskademux, which indicates that the format for the pads now reads video_%u. Note that some documentation pages of gstreamer are not updated to reflect that.
The first command, MKV to MKV should read:
gst-launch-1.0 filesrc location=test.mkv ! matroskademux name=demux \
matroskamux name=mux ! filesink location=test2.mkv \
demux.video_0 ! queue ! mux.video_0 \
demux.audio_0 ! queue ! mux.audio_0
(Note the added queues)
The second command, MKV to MKV reencoding audio should read:
gst-launch-1.0 -v filesrc location=test.mkv ! matroskademux name=demux \
matroskamux name=mux ! filesink location=test2.mkv \
demux.video_0 ! queue ! 'video/x-h264' ! h264parse ! mux. \
demux.audio_0 ! queue ! rawaudioparse ! audioconvert ! audioresample ! avenc_aac ! mux.
The same result could have been achieved by not specifying the pads and using cap filters if needed.
Thanks go to user Florian Zwoch for providing a working pipeline.
I try to write a GStreamer pipeline to capture the screen, put a box on the corner capturing the webcam and record audio (all at the same time).
If I hit Ctrl+C to stop after ten seconds, for example, I realize I only record about 2 seconds of video (and audio). Actually, I don't care that the recording were done in real time, but I just want that GStreamer records the full lenght it should be.
This is the pipeline I have so far:
gst-launch-1.0 --gst-debug=3 ximagesrc use-damage=0 \
! video/x-raw,width=1366,height=768,framerate=30/1 ! videoconvert \
! videomixer name=mix sink_0::alpha=1 sink_1::alpha=1 sink_1::xpos=1046 sink_1::ypos=528 \
! videoconvert ! queue max-size-buffers=0 max-size-time=0 max-size-bytes=0 \
! vp8enc ! webmmux name=mux ! filesink location="out.webm" \
pulsesrc ! audioconvert ! queue max-size-buffers=0 max-size-time=0 max-size-bytes=0 ! vorbisenc ! mux. \
v4l2src do-timestamp=true ! video/x-raw,width=320,height=240,framerate=30/1 ! mix.
I hope to have a solution, thank you.
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.`