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.
Related
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'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 using the following pipeline to convert an flv file to mp4.
gst-launch-1.0 -vvv -e filesrc location="c.flv" ! flvdemux name=demux \
demux.audio ! queue ! decodebin ! audioconvert ! faac bitrate=32000 ! mux. \
demux.video ! queue ! decodebin ! videoconvert ! video/x-raw,format=I420 ! x264enc speed-preset=superfast tune=zerolatency psy-tune=grain sync-lookahead=5 bitrate=480 key-int-max=50 ref=2 ! mux. \
mp4mux name=mux ! filesink location="c.mp4"
The problem is when (for example) audio is missing, the pipeline gets stuck. (Same thing happens if just hooking a fakesink to demux.audio).
I need a way for the filters to ignore missing tracks, or produce empty tracks.
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.`
I have been working on an application where I use rtspsrc to gather audio and video from one network camera to another. However I can not watch the stream from the camera and thereby cant verify that the stream works as intended. To verify that the stream is correct I want to record it on a SD card and then play the file on a computer. The problem is that I want the camera to do as much of the parsing, decoding, depayloading as possible since that is the purpose of the application.
I thereby have to separate the audio and video streams by a demuxer and do the parsing, decoding etc and thereafter mux them back into a matroska file.
The video decoder has been omitted since it is not done yet for this camera.
Demux to live playback sink(works)
gst-launch-0.10 -v rtspsrc location="rtsp://host:pass#192.168.0.91/XXX/XXXX?resolution=1280x720&audio=1&audiocodec=g711&audiosamplerate=8000&audiobitrate=64000" latency=0 name=d d. ! rtppcmudepay ! mulawdec ! audioresample ! audioconvert ! autoaudiosink d. ! rtph264depay ! ffdec_h264 ! queue ! ffmpegcolorspace ! autovideosink
Multiple rtspsrc to matroska(works)
gst-launch-1.0 -v rtspsrc location="rtsp://host:pass#192.168.0.91/XXX/XXXX?audio=1&audiocodec=g711&audiosamplerate=8000&audiobitrate=64000" latency=0 ! rtppcmudepay ! mulawdec ! audioresample ! audioconvert ! queue ! matroskamux name=mux ! filesink location=/var/spool/storage/SD_DISK/testmovie.mkv rtspsrc location="rtsp://root:pass#192.168.0.91/axis-media/media.amp?resolution=1280x720" latency=0 ! rtph264depay ! h264parse ! mux.
Single rtspsrc to matroska(fails)
gst-launch-1.0 -v rtspsrc location="rtsp://host:pass#192.168.0.91/XXX/XXXX?resolution=1280x720&audio=1&audiocodec=g711&audiosamplerate=8000&audiobitrate=64000" latency=0 name=d d. ! queue ! rtppcmudepay ! mulawdec ! audioresample ! audioconvert ! queue ! matroskamux name=mux d. ! queue ! rtph264depay ! h264parse ! queue ! mux. ! filesink location=/var/spool/storage/SD_DISK/testmoviesinglertsp.mkv
The last example fails with the error message
WARNING: erroneous pipeline: link without source element
Have i missunderstood the usage of matroska mux and why does the 2 above examples work but not the last?
The problem is here:
queue ! mux. ! filesink
You need to do
queue ! mux. mux. ! filesink
mux. means that gst-launch should select a pad automatically from mux. and link it. You could also specify manually a name, like mux.src. So syntactically you are missing another element/pad there to link to the other element.