I am trying to figure out the proper gstreamer element to use to transmit AAC audio over RTP.
By dumping the dot graph of a playbin on the file I can conclude that the caps coming out of the tsdemux is audio/mpeg,mpegversion:2,stream-format:adts .
If I use the following pipeline
gst-launch-1.0 -v filesrc location=$BA ! tsdemux ! audio/mpeg ! rtpmpapay ! filesink location=/tmp/test.rtp
it fails:
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
/GstPipeline:pipeline0/GstCapsFilter:capsfilter1: caps = audio/mpeg
WARNING: from element /GstPipeline:pipeline0/GstTSDemux:tsdemux0: Delayed linking failed.
Additional debug info:
/var/tmp/portage/media-libs/gstreamer-1.12.3/work/gstreamer-1.12.3/gst/parse/grammar.y(510): gst_parse_no_more_pads (): /GstPipeline:pipeline0/GstTSDemux:tsdemux0:
failed delayed linking some pad of GstTSDemux named tsdemux0 to some pad of GstRtpMPAPay named rtpmpapay0
ERROR: from element /GstPipeline:pipeline0/GstTSDemux:tsdemux0: Internal data stream error.
Additional debug info:
/var/tmp/portage/media-libs/gst-plugins-bad-1.12.3/work/gst-plugins-bad-1.12.3/gst/mpegtsdemux/mpegtsbase.c(1613): mpegts_base_loop (): /GstPipeline:pipeline0/GstTSDemux:tsdemux0:
streaming stopped, reason not-linked (-1)
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
Freeing pipeline ...
Which gstreamer element should I be using to wrap AAC audio in an RTP packet?
I guess its rtpmp4apay: RTP MPEG4 audio payloader. Maybe you want/need aacparse before the payloader as well.
Related
I'm using this Gstreamer pipeline to send an RTSP stream of a camera.
./gst-rtsp-launch --port 8554 "( v4l2src device=/dev/video0 ! video/x-raw,framerate=30/1,width=640,height=480 ! rtpvrawpay name=pay0 pt=96 )"
I want to using playbin, so I don't need to specify the type of video from the rtsp stream. If I use this pipeline, I can get a single image from the camera:
gst-launch-1.0 playbin uri=rtsp://(ip-of-camera):8554 video-sink="jpegenc ! filesink location=capture1.jpeg"
But if I try this pipeline, to save as a file:
gst-launch-1.0 playbin uri=rtsp://(ip-of-camera):8554 video-sink="videoconvert ! video/x-h264,width=320,height=240 ! mp4mux ! filesink location=test.mp4"
I get this error:
ERROR: from element /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0/GstRTSPSrc:source/GstUDPSrc:udpsrc1: Internal data stream error.
Additional information for debugging:
../libs/gst/base/gstbasesrc.c(3127): gst_base_src_loop (): /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0/GstRTSPSrc:source/GstUDPSrc:udpsrc1:
streaming stopped, reason not-negotiated (-4)
Execution ended after 0:00:00.094968753
Defining the processing queue to NULL...
Freeing the processing queue...
Note: I had translate the last two lines.
Is there a problem in the pipeline I'm using to save the stream as a file?
I copied your pipeline and in worked.
I only added -e after gst-launch-1.0, so that GStreamer writes all necessary information to the file when ctrl-C is pressed:
gst-launch-1.0 -e playbin uri=(rtsp url) video-sink="videoconvert ! video/x-h264,width=320,height=240 ! mp4mux ! filesink location=test.mp4"
Maybe you are using an older version of GStreamer?
I am using GStreamer 1.20.0
Initial Issue
I would like to use gstreamer plugin opusdec to decode an Opus bitstream. The final purpose is to make glue around it with appsrc and appsink as input/output, to decode 20 ms Opus packets coming from a RTP packet payload and provide PCM sample.
Remark: I can't use gstreamer rtpopusdepay
The following pipeline works:
gst-launch-1.0 filesrc location=testvector01.bit.opus ! oggdemux !
opusdec ! fakesink
In my final application I'm no expected OGG contained data so I did the following:
1) Desencapsulate Opus bitstream
gst-launch-1.0 filesrc location=testvector01.bit.opus ! oggdemux !
filesink location = testvector01.bit.demux
That works. And then:
2) Decode Opus bitstream
gst-launch-1.0 filesrc location=testvector01.bit.demux ! opusdec !
fakesink
and I have the following error:
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
ERROR: from element /GstPipeline:pipeline0/GstFileSrc:filesrc0: Internal data flow error.
Additional debug info:
gstbasesrc.c(2865): gst_base_src_loop (): /GstPipeline:pipeline0/GstFileSrc:filesrc0:
streaming task paused, reason error (-5)
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
Freeing pipeline ..
Input File
testvector01.bit.opus
From Opus test vector :https://people.xiph.org/~greg/opus_testvectors/
My question is:
What is the proper way to use gstreamer plugin opusec without transport container?
Update
Gstreamer version 1.2.4
As recommended I tried to add opusparse after filesrc and got the following error.
Pipeline is PREROLLING ...
(gst-launch-1.0:5147): GStreamer-WARNING **:
gstpad.c:4555:store_sticky_event:<opusparse0:src> Sticky event
misordering, got 'caps' before 'stream-start'
(gst-launch-1.0:5147): GStreamer-WARNING **:
gstpad.c:4555:store_sticky_event:<opusdec0:sink> Sticky event
misordering, got 'caps' before 'stream-start' Pipeline is PREROLLED
... Setting pipeline to PLAYING ... New clock: GstAudioSinkClock
ERROR: from element /GstPipeline:pipeline0/GstOpusDec:opusdec0:
Decoding error: -4 Additional debug info: gstopusdec.c(460):
opus_dec_chain_parse_data ():
/GstPipeline:pipeline0/GstOpusDec:opusdec0 Execution ended after
0:00:00.063372478 Setting pipeline to PAUSED ... Setting pipeline to
READY ... Setting pipeline to NULL ... Freeing pipeline ...
GStreamer 1.8.1
The following pipeline
gst-launch-1.0 filesrc location = testvector01.bit.demux ! opusparse !
opusdec ! audioconvert ! alsasink
halt here:
Setting pipeline to PAUSED ... Pipeline is PREROLLING ...
Gstreamer 1.13.1
gst-launch-1.0 filesrc location = testvector01.bit.demux ! opusparse !
opusdec ! alsasink
Playback just produce a short audio glitch while no gstreamer error is raised.
gst-launch-1.0 filesrc location = testvector01.bit.opus ! oggdemux ! opusparse !
opusdec ! alsasink
Playback is choppy while no gstreamer error is raised.
Regards,
appsrc is-live=true do-timestamp=true name=audiosrc ! opusparse ! oggmux ! filesink location=test.ogg
gstreamer 1.14.1 works fine
You need to have a parser (opusparse) in between as opusdec doesn’t know what format it is, try the following pipeline:
gst-launch-1.0 filesrc location=testvector01.bit.demux ! opusparse !
opusdec ! fakesink dump=true
I am trying to encode uncompressed video in H.265; however, when I write the following pipeline I receive an error message that I cannot resolve. I am following the example code in Tegra X1 Multimedia User Guide, and I do not understand why the following pipeline does not work. I am a beginner in video compression so any help would be very useful. The code/error message:
ubuntu#tegra-ubuntu:~$ gst-launch-1.0 filesrc location=small_mem_vid.mov ! 'video/x-raw, format=(string)I420, framerate=(fraction)30/1, width=(int)1280, height=(int)720' ! omxh265enc ! filesink location=new_encode.mov -e
Setting pipeline to PAUSED ...
Inside NvxLiteH264DecoderLowLatencyInitNvxLiteH264DecoderLowLatencyInit set DPB and MjstreamingInside NvxLiteH265DecoderLowLatencyInitNvxLiteH265DecoderLowLatencyInit set DPB and MjstreamingPipeline is PREROLLING ...
Framerate set to : 30 at NvxVideoEncoderSetParameterNvMMLiteOpen : Block : BlockType = 8
===== MSENC =====
NvMMLiteBlockCreate : Block : BlockType = 8
ERROR: from element /GstPipeline:pipeline0/GstOMXH265Enc-omxh265enc:omxh265enc-omxh265enc0: Could not write to resource.
Additional debug info:
/dvs/git/dirty/git-master_linux/external/gstreamer/gst-omx/omx/gstomxvideoenc.c(2139): gst_omx_video_enc_handle_frame (): /GstPipeline:pipeline0/GstOMXH265Enc-omxh265enc:omxh265enc-omxh265enc0:
Failed to write input into the OpenMAX buffer
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
Freeing pipeline ...
ubuntu#tegra-ubuntu:~$
Are you sure the .mov file is really uncompressed video? The .mov extension is commonly used for quicktime video. You could use "mediainfo" in Linux to discover more details about the format of the file. In that case I don't think you can go directly from filesrc to the encoder. You probably need a qtdemux and a decoder, maybe avdec_h264 depending on what mediainfo shows.
You also might want to enable some more detailed debugging:
export GST_DEBUG=*:4
I am trying to user gst-launch to stream mp3 audio over tcp, this is what I am trying :
$ gst-launch-0.10 filesrc location="/path/to/file.mp3" ! tcpserversink host=0.0.0.0 port=3000
but it doesn't work the output is as follow :
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
ERROR: from element /GstPipeline:pipeline0/GstTCPServerSink:tcpserversink0: Internal GStreamer error: negotiation problem. Please file a bug at http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer.
Additional debug info:
gstmultifdsink.c(2700): gst_multi_fd_sink_render (): /GstPipeline:pipeline0/GstTCPServerSink:tcpserversink0:
Received first buffer without caps set
Execution ended after 94657 ns.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ..
Whats the problem where I am going wrong?
I did lots of search on internet but didn't found the right document how to use gst-launch properly. if anyone can please point me to right doc or tell me how to use it it will great.
tcpserversink complains about missing caps on its sink pad:
Received first buffer without caps set
This is because tcpserversink wants to know what it sends.
One way to tell it would be to manually decode and re-encode the stream:
gst-launch-0.10 filesrc location="/path/to/file.mp3" ! mad ! audioconvert ! lame ! tcpserversink host=0.0.0.0 port=3000
But that's just waste of CPU power.
There's an element called mpegaudioparse that (apart from some other stuff) figures out the details of the mpeg stream and sets its output caps accordingly. By simply putting it between your filesrc and the tcpserversink you'll end up with a working pipeline:
$ gst-launch-0.10 filesrc location="/path/to/file.mp3" ! mpegaudioparse ! tcpserversink host=0.0.0.0 port=3000
I'm trying to play a local mpeg2 TS file with gstreamer with this:
gst-launch filesrc location=open_season.mpg ! mpeg2dec ! xvimagesink
The first frame appears as big blocks of color and then stops. Any thoughts about what I'm doing wrong here? Does a -TS file need to be handled differently than this?
Here's the log:
$ gst-launch filesrc location=open_season.mpg ! mpeg2dec ! xvimagesink
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ....
WARNING: from element /GstPipeline:pipeline0/GstXvImageSink:xvimagesink0: Internal data flow problem..
Additional debug info:.
gstbasesink.c(3492): gst_base_sink_chain_unlocked (): /GstPipeline:pipeline0/GstXvImageSink:xvimagesink0:
Received buffer without a new-segment. Assuming timestamps start from 0.
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
Got EOS from element "pipeline0".
Execution ended after 6866757291 ns.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ..
I think first you should first try to play the file with the help of playbin2. If you are able to play it then u should use decodebin2 ,debug its output and construct your pipeline accordingly.
The syntax for playbin2 is as follows :-
gst-launch playbin2 uri = file:///home/user1031040/Desktop/file.mpg
The syntax for decodebin2 is as follows:-
gst-launch filesrc location = file.mpg ! decodebin2 ! autovideosink