I am new using g streamer, and i try to use the emit-stats properties in tsdemux
How can I do to use this in my pipeline?
I'm trying to get the program clock reference value of a signal transport stream but no way to get it.
Properties in GStreamer are normally accessed by using the normal GLib APIs : g_object_set and g_object_get. Doing g_object_set (v1_demux, "emit-stats", TRUE, NULL);, supposing that v1_demux is a GstTSDemux*, will start emitting messages containing the PTS and DTS of the packets that flow into the demuxer.
Element messages in GStreamer are emitted by gst_element_post_message. In order to receive them in your application, it needs to set up a bus watch on the main pipeline's GstBus.
Just for the record, you can test how the property works and see the content of the messages by running this example pipeline in gst-launch :
gst-launch-1.0 -m filesrc location="$YOUR_TRANSPORT_STREAM" ! tsdemux emit-stats=1 ! fakesink
Running this with one of the transport streams on my HDD, I can see messages with the PTS and DTS being emitted from the demuxer element :
Got message #77 from element "tsdemux0" (element): tsdemux, pid=(uint)1803, offset=(guint64)266020, pts=(guint64)8429319339;
Got message #78 from element "tsdemux0" (element): tsdemux, pid=(uint)1805, offset=(guint64)273540, pts=(guint64)8429311261;
Got message #79 from element "tsdemux0" (element): tsdemux, pid=(uint)1802, offset=(guint64)282564, dts=(guint64)8429444461;
However, it doesn't look like PCR and OPCR values are emitted. You'll have to add this functionality yourself.
Thanks for the info.
was test commands and see the script and check the values, but is costing me add messages emit-stats in my line.
If I created a bus watch on the main pipeline's GstBus,to see the video duration and as playtime in my line, but can not see messages stats and video simultaneously. I still investigating comohacerlo as storing information pts and dts in some way.
My idea is get stamps of two videos and subtract this to calculate an automatic offset in one video.
Related
I'm using appsrc to generate an HLS stream, this is my successful pipeline:
appsrc->videoconvert->openh264enc->h264parse->mpegtsmux->hlssink
However, I'd like to generate some audio via audiotestsrc before mpegtsmux which would look like the following:
audiotestsrc->lamemp3enc->mpegtsmux
Audiotestsrc and lame have 'always' pads, so I link the two just like my other video elements.
When it comes to linking lame's "always" "src", to mpegtsmux's "request" "sink_%d", the result says that there's no issue:
//Returns 0
gst_pad_link(h264ParsePad, mpegtsmuxSinkPad);
//Returns 0
gst_pad_link(audioEncPad, mpegtsmuxSinkPad);
//Returns 0
gst_pad_link(mpegtsmuxSrcPad, hlssinkPad);
But running the app results in pipeline failure with
"Internal data stream error."
Removing the audioEncPad linking just makes the stream work like normal but of course without audio. How should I go about doing this?
Few things needed to be done:
Use aacparse
Clean the solution
Link voaacenc with aacparse
#2 caused me a lot of torment since everything theoretically should've worked. D'oh.
The goal here is to be able to send some metadata (timestamps, objects found per frame) with a stream within a single pipeline or multiple pipelines over network (e.g. RTP UDP).
Within The pipeline:
This is straight forward by defining a new GstMeta API and register and implement it. Then add it to GstBuffers via buffer probes or another element solely designed for this purpose.
Multiple pipelines over Network:
The only solution I have heard about is to add one or more RTP header extension to RTP packets coming out of a payloader. Actually some people decided to transform their Custom GstMeta into those header extensions. But Is This really a good use-case for it?
There's more ....
Another idea is to create your own custom media type video/x-mytype and for that you will need to write a typefinding function and an autoplugger on top of it, Not to mention a couple of elements to deal with this new type and convert it to other exiting media types. Now that new type should actually have a place where it can handle the meta data I was talking about.
That summarizes my research, Are there any other methods that I am not aware of?
I would definitely appreciate your input on this!
Thanks in advance!
I decided to implement a new media type inside gstreamer (e.g. video/x-h265-with-meta). This new media type shall have inherent metadata handling capabilities. You can design how it encode/decode the metadata as you see fit in this new type. You would also need write 2 more gstreamer elements. one to convert regular video/x-h265 to video/x-h265-with-meta (say muxmymeta) and the other to reverse such conversion (say demuxmymeta). So essentially muxmymeta will read the metadata injected in the GstBuffer* via GstMeta API implementation and encodes in the outgoing stream. demuxmymeta will decode such metadata from the stream and add it back to GstBuffer*. To ensure that for every encoded h265 frame will have a corresponding meta, muxmymeta must have its src caps in this format:
video/x-h265
stream-format: byte-stream
alignment: au
The byte-stream is mandatory for network transmission and the au alignment forces a whole encoded frame as input for muxmymeta.
I also used rtpgstpay and rtpgstdepay elements to use this new media type with RTP/RTSP. You will NOT need to write a typefinding function at this point since rtpgstpay/rtpgstdepay will be able to figure that out from the caps. This works well out of the box for h265 streams in gstreamer v1.6.2 but I had to make a fix inside gstreamer itself to make it work with h264 as well. Finally, to make autopluggers such as decodebin/uridecodebin to decode your stream, you must set the klass for demuxmymeta to Codec/Decoder/Demuxer and set a priority for it other than none, this basically tells gstreamer to use this element for decoding.
Here are 2 pipelines for illustration:
Sending H265 RTSP with metadata pipeline (set as the launch line for GstRTSPMediaFactory)
( rtspsrc location=rtsp://localhost:8554/test latency=0 ! rtph265depay ! h265parse ! muxmymeta ! rtpgstpay name=pay0 pt=96 )
Here, the rtsp://localhost:8554/test stream is a regular h265 stream video/x-h265. yet assume that the rtsp-server will produce rtsp://localhost:8553/test-meta which has the type video/x-h265-with-meta
Receiver
rtspsrc location=rtsp://localhost:8553/test-meta ! rtpgstdepay ! demuxmymeta ! h265parse ! avdec_h265 ! videoconvert ! autovideosink
With an Autoplugger it will look as neat as:
uridecodebin uri=rtsp://localhost:8553/test-meta ! autovideosink
Though this might seem like a longshot solution, it provides a way to transport self-synchronizing metadata between pipelines over the network.
how can I connect "qtdemux" and "ffdec_h264" using c code with Gstreamer ?
demux= gst_element_factory_make("qtdemux", "demux");
dec = gst_element_factory_make ("ffdec_h264", "dec");
I believe you are using 0.10.x? Gst 1.x doesn't have ffmpeg elements.
Here are the steps:
You will have to attach a callback to "pad-added" signal of the qtdemux.
The callback will be invoked from the qtdemux for every stream
(eg. audio,video) present in the source.
The pad connection will have to be done inside the callback.
Go to page 32 on
https://lzy-s-prct.googlecode.com/files/gstreamer-manual.pdf
for the example.
I'm looking for the correct technique, if one exists, for dynamically replacing an element in a running gstreamer pipeline. I have a gstreamer based c++ app and the pipeline it creates looks like this (using gst-launch syntax) :
souphttpsrc location="http://localhost/local.ts" ! mpegtsdemux name=d ! queue ! mpeg2dec ! xvimagesink d. ! queue ! a52dec ! pulsesink
During the middle of playback (i.e. GST_STATE_PLAYING is the pipeline state and the user is happily watching video), I need to remove souphttpsrc from the pipeline and create a new souphttpsrc, or even a new neonhttpsource, and then immediately add that back into the pipeline and continue playback of the same uri source stream at the same time position where playback was before we performed this operation. The user might see a small delay and that is fine.
We've barely figured out how to remove and replace the source, and we need more understanding. Here's our best attempt thus far:
gst_element_unlink(source, demuxer);
gst_element_set_state(source, GST_STATE_NULL);
gst_bin_remove(GST_BIN(pipeline), source);
source = gst_element_factory_make("souphttpsrc", "src");
g_object_set(G_OBJECT(source), "location", url, NULL);
gst_bin_add(GST_BIN(pipeline), source);
gst_element_link(source, demuxer);
gst_element_sync_state_with_parent(source);
This doesn't work perfectly because the source is playing back from the beginning and the rest of the pipeline is waiting for the correct timestamped buffers (I assume) because after several seconds, playback picks back up. I tried seeking the source in multiple ways but nothing has worked.
I need to know the correct way to do this. It would be nice to know a general technique, if one exists, as well, in case we wanted to dynamically replace the decoder or some other element.
thanks
I think this may be what you are looking for:
http://cgit.freedesktop.org/gstreamer/gstreamer/tree/docs/design/part-block.txt
(starting at line 115)
I'm constructing a gstreamer pipeline that receives two RTP streams from an networked source:
ILBC Audio stream + corresponding RTCP stream
H263 Video stream + corresponding RTCP stream
Everything is put into one gstreamer pipeline so it will use the RTCP from both streams to synchronize audio/video. So far I've come up with this (using gst-launch for prototyping):
gst-launch -vvv gstrtpbin name=rtpbin
udpsrc caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H263-2000" port=40000 ! rtpbin.recv_rtp_sink_0
rtpbin. ! rtph263pdepay ! ffdec_h263 ! xvimagesink
udpsrc port=40001 ! rtpbin.recv_rtcp_sink_0
rtpbin.send_rtcp_src_0 ! udpsink port=40002 sync=false async=false
udpsrc caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMU,encoding-params=(string)1,octet-align=(string)1" port=60000 rtpbin.recv_rtp_sink_1
rtpbin. ! rtppcmudepay ! autoaudiosink
udpsrc port=60001 ! rtpbin.recv_rtcp_sink_1
rtpbin.send_rtcp_src_1 ! udpsink port=60002 sync=false async=false
This pipeline works well if the networked source starts out with sending both video and audio. If the videostream is paused later on, gstreamer will still playback audio and even will start playing back the video when the networked source resumes the video stream.
My problem is however that if the networked source starts out with only an audio stream (video might be added later on), the pipeline seems to pause/freeze until the video stream starts as well.
Since video is optional (and can be added/removed at will by the user) in my application, is there any way I can hook up for instance an 'videotestsrc' that will provide some kind of fallback video data to keep the pipeline running when there is no networked video data?
I've tried experimenting with 'videotestsrc' and a thing called 'videomixer' but I think that mixer still requires both streams to be alive. Any feedback is greatly appreciated!
I present a simple function for pause resume by changing bins. In the following example I provide the logic to change destination bin on the fly dynamically. This shall not completely stop the pipeline which is what you seek I believe. A similar logic could be used for src bins. Here you may remove your network source bin and related decoder/demux bins and add videotestsrc bins.
private static void dynamic_bin_replacement(Pipeline pipe, Element src_bin, Element dst_bin_new, Element dst_bin_old) {
pipe.pause();
src_bin.unlink(dst_bin_old);
pipe.remove(dst_bin_old);
pipe.add(dst_bin_new);
dst_bin_new.syncStateWithParent();
src_bin.link(dst_bin_new);
pipe.ready();
pipe.play();
}
The other logic you may want to try is "PADLOCKING". Please take a look at the following posts
http://cgit.freedesktop.org/gstreamer/gstreamer/tree/docs/design/part-block.txt
and
http://web.archiveorange.com/archive/v/8yxpz7FmOlGqxVYtkPb4
and
Adding and removing audio sources to/from GStreamer pipeline on-the-go
UPDATE
Try output-selector and input-selector bins as they seem to be better alternative. I found them most reliable and have had immense luck with them. I use fakesink or fakesrc respectively as the other end of the selector.
valve bin is another alternative that I found doesn't even need fakesink or fakesrc bins. It is also extremely reliable.
Also the correct state transition order for media file source
NULL -> READY -> PAUSED -> PLAYING (Upwards)
PLAYING -> PAUSED -> READY -> NULL (Downwards)
My order in the above example should be corrected where ready() should come before pause(). Also I would tend to think un-linking should be performed after null() state and not after pause(). I haven't tried these changes but theoretically they should work.
See the following link for detailed info
http://cgit.freedesktop.org/gstreamer/gstreamer/tree/docs/design/part-states.txt?h=BRANCH-RELEASE-0_10_19