I'm trying to use MEncoder to dump an RTSP stream I get. When I try to do the same, I get an error saying "librtsp: buffer overflow in rtsp_get".
The full console output is given below.
C:\mplayer-svn-37594>mplayer -dumpstream "rtsp://qa.vibrnet.com:80/mov/video.sav?MAC=00C0021F1116&channel=2&GUID=betauser"
MPlayer Redxii-SVN-r37594-4.9.3 (i686) (C) 2000-2016 MPlayer Team
FFmpeg version: N-77758-g6e24946
Build date: 2016-01-09 00:49:37 EST
Playing rtsp://qa.vibrnet.com:80/mov/video.sav?MAC=00C0021F1116&channel=2&GUID=betauser.
Resolving qa.vibrnet.com for AF_INET...
Connecting to server qa.vibrnet.com[204.x.x.x]: 80...
librtsp: buffer overflow in rtsp_get
When I tried to play the stream, I get this error as well. What could be the reason? Help me resolve this one. Thank you!
Related
I'm trying to get fmp4 HLS playing back on a new Chromecast (3rd gen I believe, not Ultra).
I've tried encoding the content with ffmpeg using both x264 and h264 libraries.
The main profile initially gives me a codec not supported error, remove the codec list from the hls manifest fixes this issue.
Switching to baseline (which is not ideal) doesn't give the codec error.
Both then (after removing the codec definitions or using baseline) give the following error:
Uncaught Error: Unable to derive timescale
at Xl (www.gstatic.com/cast/sdk/libs/mediaplayer/1.0.0/media_player.js:344)
at Y.$e (www.gstatic.com/cast/sdk/libs/mediaplayer/1.0.0/media_player.js:337)
at Y.k.processSegment (www.gstatic.com/cast/sdk/libs/mediaplayer/1.0.0/media_player.js:340)
at Am.k.processSegment (www.gstatic.com/cast/sdk/libs/mediaplayer/1.0.0/media_player.js:384)
at Mj.$e (www.gstatic.com/cast/sdk/libs/mediaplayer/1.0.0/media_player.js:238)
at Wj (www.gstatic.com/cast/sdk/libs/mediaplayer/1.0.0/media_player.js:236)
at Oj (www.gstatic.com/cast/sdk/libs/mediaplayer/1.0.0/media_player.js:240)
at Mj.fd (www.gstatic.com/cast/sdk/libs/mediaplayer/1.0.0/media_player.js:239)
at Nc (www.gstatic.com/cast/sdk/libs/mediaplayer/1.0.0/media_player.js:39)
at wi.Mc.dispatchEvent (www.gstatic.com/cast/sdk/libs/mediaplayer/1.0.0/media_player.js:38)
Make sure you're not setting:
loadRequestData.media.hlsSegmentFormat
For TS I had to set:
loadRequestData.media.hlsSegmentFormat = cast.framework.messages.HlsSegmentFormat.TS;
But for fmp4 I commented this out.
I need a RTSP-server that can listen on a configured port (8554 for example), and then, for example, if I run FFmpeg with:
ffmpeg -f v4l2 -i /dev/video0 -c:v libx264 -intra -an -f rtsp -rtsp_transport tcp rtsp://192.168.1.10:8554/test
Then the RTSP-server will RECORD the video, and to play it, I just need to run it with:
ffplay -i rtsp://192.168.1.10:8554/test
I need the RTSP-server to support TCP transport and H264 video encoder and OPUS audio encoder and stream from a live-video (not from a file) + the program should be unlicensed.
This server works great, but don't support OPUS.
Live555 support H264 and OPUS, but only streams from files (VOD).
I've have found some other servers that can stream directly from /dev/video0, but it's also not a good solution for me.
Wowza and Red5Pro does answer all the above requirements, except that they are licenced programs.
Any suggestions for a RTSP-server that support all the above requirements?
EDIT:
I've tried Gstreamer and it looks promising, but I still didn't success.
However, I'm quite sure I'm on the right way (perhaps I don't know how to use yet the pipelines).
I've built gst-rtsp-server, version 1.13.91.
Then, I ran ./test-record "( decodebin name=depay0 ! videoconvert ! rtspsink )"
I ran netstat -anp and I can see clearly, the server is listening on tcp port 8554.
Now it's time to stream to server. I've tried it once with Gstreamer and once with FFmpeg.
Gstreamer
gst-launch-1.0 videotestsrc ! x264enc ! rtspclientsink location=rtsp://127.0.0.1:8554/test
FFmpeg
ffmpeg -f v4l2 -video_size 640x480 -i /dev/video0 -c:v libx264 -qp 10 -an -f rtsp -rtsp_transport tcp rtsp://127.0.0.1:8554/test
In both cases, I can see the RTP packets in wireshark,
and by calling again to netstat -anp, I see:
tcp 0 0 0.0.0.0:8554 0.0.0.0:* LISTEN 14386/test-record
tcp 0 0 127.0.0.1:8554 127.0.0.1:46754 ESTABLISHED 14386/test-record
tcp 0 0 127.0.0.1:46754 127.0.0.1:8554 ESTABLISHED 19479/ffmpeg
So I can surly understand that I'm streaming (or streaming something...). However, when I'm trying to play the video, I'm getting failure (I've tried to play with Gstreamer, FFplay and VLC - all fails...):
Gstreamer
gst-launch-1.0 rtspsrc location=rtsp://127.0.0.1:8554/test latency=300 ! decodebin ! autovideoconvert ! autovideosink
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Progress: (open) Opening Stream
Progress: (connect) Connecting to rtsp://127.0.0.1:8554/test
Progress: (open) Retrieving server options
Progress: (open) Retrieving media info
ERROR: from element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0: Could not get/set settings from/on resource.
Additional debug info:
gstrtspsrc.c(7507): gst_rtspsrc_retrieve_sdp (): /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0:
Server can not provide an SDP.
ERROR: pipeline doesn't want to preroll.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...
FFplay
ffplay -i rtsp://127.0.0.1:8554/test
[rtsp # 0x7fb140000b80] method DESCRIBE failed: 405 Method Not Allowed
rtsp://127.0.0.1:8554/test: Server returned 4XX Client Error, but not one of 40{0,1,3,4}
VLC
vlc rtsp://127.0.0.1:8554/test
VLC media player 3.0.8 Vetinari (revision 3.0.8-0-gf350b6b)
[0000000000857f10] main libvlc: Running vlc with the default interface. Use 'cvlc' to use vlc without interface.
Qt: Session management error: None of the authentication protocols specified are supported
[00007f9fdc000ea0] live555 demux error: Failed to connect with rtsp://127.0.0.1:8554/test
[00007f9fdc001d10] satip stream error: Failed to setup RTSP session
Any ideas what I'm doing wrong ?
Wowza SE works with H264, Opus, VP8 as it supports WebRTC.
This plugin provides a turnkey setup for broadcasting channels live with WebRTC, RTMP, RTSP trough Wowza SE. Also can handle all stream types including RTSP with FFMPEG for on demand adaptive transcoding (in example between WebRTC & HLS).
https://wordpress.org/plugins/videowhisper-live-streaming-integration/
Well, the closest RTSP-server I found so far that matches (almost) all my requirements can be found here: https://github.com/RSATom/RtspRestreamServer (credits for the RTSP-server are for RSATom).
Here is the checklist for all of the features I was looking for:
Support TCP transpot.
Support H264 video codec (currently hard-codec for this codec only).
Support OPUS audio codec (not supported yet, but the server is based Gstreamer library, so it has all the infrastructure to support all the codecs Gstreamer supports - I just need to update the code and make it more generic).
Support RTSP RECORD option from a client with a Live-Stream.
Support RTSP PLAY option from a client.
URL and PORT should be configurable (currently hard-codec - just need to update the code and make it more generic).
The server is Unlicensed.
I'm trying to convert an WMA file into mp4 in order to upload the file to youtube.
VN550672.wma.zip
Although the conversion is successful (see below) i'm not able to upload the file to youtube. I'm getting the following error
The video has failed to process. Please make sure you are uploading a supported file type.
VN550672.mp4.zip
Any suggestions?
System configuration:
Python version: 3.6.3
Pydub version: 0.22.1
ffmpeg or avlib?: ffmpeg
ffmpeg/avlib version: 2.8.4
Could it be as easy as youtube requires that the media file includes a video stream? The wma file only has a audio stream.
You can try to transcode and add a dummy video stream using
ffmpeg -i VN550672.WMA -f lavfi -i color=size=426x240 VN550672.mp4
(426x240 is the youtube suggested minimal resolution)
I am getting the following error when trying to convert an mp4 video to gif using gifify
Unable to find application named 'Cloud'
Does anyone know how to debug or investigate this type of issue?
Appears that gifify "by default" attempts to upload the gif to "CloudApp" ("Cloud" in earlier versions) https://github.com/jclem/gifify/blob/master/gifify.sh#L88
So run it like gifify -n movie.mp4 instead.
I suggest you add a new issue telling them to log better when there is no CloudApp installed here: https://github.com/jclem/gifify/issues
I'm trying to create a very simply Gstreamer pipeline where I have a source element that is my FaceTime camera and a sink element that is a udp sink.
I first install Gstreamer using the instructions here. I ran some of the basic pipelines no problem; however, when I tried to use the following command
./gst-launch-0.10 v4l2src ! xviimagesink
I got the following error:
ERROR: pipeline could not be constructed: no element "v4l2src".
So I did some digging and turns out that the v4l2src plugin is in a gst-plugins-good. I installed these good plugins using macports using the following command:
port install gst-plugins-good
After a very long time everything installed without error. Now gst-launch appears in three places.
/Library/Frameworks/GStreamer.framework/Versions/0.10/bin/gst-launch-0.10
/opt/local/bin/gst-launch
/opt/local/bin/gst-launch-0.10
If I try to run the above mentioned pipline from any of those directories I still get
ERROR: pipeline could not be constructed: no element "v4l2src".
If I type the following command from anywhere I get some more errors but seems like it still is not finding v4lsrc.
gst-launch v4l2src ! xviimagesink
Gives:
Dynamic session lookup supported but failed: launchd did not provide a socket path, verify that org.freedesktop.dbus-session.plist is loaded!
Dynamic session lookup supported but failed: launchd did not provide a socket path, verify that org.freedesktop.dbus-session.plist is loaded!
Dynamic session lookup supported but failed: launchd did not provide a socket path, verify that org.freedesktop.dbus-session.plist is loaded!
Dynamic session lookup supported but failed: launchd did not provide a socket path, verify that org.freedesktop.dbus-session.plist is loaded!
GConf Error: Failed to contact configuration server; some possible causes are that you need to enable TCP/IP networking for ORBit, or you have stale NFS locks due to a system crash. See http://projects.gnome.org/gconf/ for information. (Details - 1: Failed to get connection to session: Not enough memory)
ERROR: pipeline could not be constructed: no element "v4l2src".
So it seems like I have GStreamer mess and I still can't get my camera to work because GStreamer can't find v4l2src.
Some help would be appreciated! Thanks in advance.
v4l2src means "video-for-linux (ver.2) source".
since you are not running "linux", it is not so surprising that you cannot use "v4l2".
you might try to use the osxvideosrc (afaik this is in gstreamer-plugins-bad).
generally i suggest to check which elements are installed on your machine when you are looking for a a specific functionality, e.g.:
$ gst-inspect | grep -i video |grep -i source
PS: and usually i find it a good idea to throw some colorspace-converter (like ffmpegcolorspace) between a video-source and and -sink.
For me those two works from MacPorts (https://www.macports.org/):
GStreamer 1.0, applemedia: avfvideosrc: Video Source (AVFoundation), use device-index parameter to select a device (index will vary depending on the connection order).
bash-3.2# port install gstreamer1*
iCeDeROM:~ cederom$ gst-inspect-1.0 |grep video | grep src
inter: intervideosrc: Internal video source
decklink: decklinkvideosrc: Decklink Video Source
applemedia: qtkitvideosrc: Video Source (QTKit)
applemedia: avfvideosrc: Video Source (AVFoundation)
ximagesrc: ximagesrc: Ximage video source
videotestsrc: videotestsrc: Video test source
autodetect: autovideosrc: Auto video source
GStreamer 0.10 (autodetect: autovideosrc: Auto video source)
bash-3.2# port install gstreamer0*
iCeDeROM:~ cederom$ gst-inspect-0.10 |grep video | grep src
ximagesrc: ximagesrc: Ximage video source
inter: intervideosrc: FIXME Long name
gsettings: gsettingsvideosrc: GSettings video src
gconfelements: gconfvideosrc: GConf video source
autodetect: autovideosrc: Auto video source
applemedia: qtkitvideosrc: Video Source (QTKit)
applemedia: miovideosrc: Video Source (MIO)
videotestsrc: videotestsrc: Video test source
I use autovideosink or osxvideosink for testing (second works faster, first use Xorg). Use gst-inspect <module> for module information.