This question is the follow up question to this thread: AR Drone 2 and ffserver + ffmpeg streaming
We are trying to get a stream from our AR Drone through a Debian server and into a flash application.
The big picture looks like this:
AR Drone --> Gstreamer --> CRTMPServer --> Flash Application
We are using the PaveParse plugin for Gstreamer found in this thread: https://projects.ardrone.org/boards/1/topics/show/4282
As seen in the thread the AR Drone is using PaVE, Parrot Video Ecapsulation, which is unrecognizable by most players like VLC. The PaVeParse plugin removes these.
We have used different pipelines and they all yield the same error.
Sample pipeline:
GST_DEBUG=3 gst-launch-0.10 tcpclientsrc host=192.168.1.1 port=5555 ! paveparse ! queue ! ffdec_h264 ! queue ! x264enc ! queue ! flvmux ! queue ! rtmpsink localtion='rtmp://0.0.0.0/live/drone --gst-plugin-path=.
The PaVEParse plugin needs to be located at the gst-plugin-path for it to work.
A sample error output from Gstreamer located in the ffdec_h264 element can be found at: http://pastebin.com/atK55QTn
The same thing will happen if the decoding is taking place in the player/dumper e.g. VLC, FFplay, RTMPDUMP.
The problem comes down to missing headers: PPS Reference is non-existent. We know that the PaVEParse plugin removes PaVE headers but we suspect that when these are removed there are no H264 headers for the decoder/player to identify the frame by.
Is it possible to "restore" these H264 headers either from scratch or by transforming the PaVE headers?
Can you please share a sample of the traffic between gstreamer and crtmpserver?
You can always use the LiveFLV support built inside crtmpserver. Here are more details:
Re-Stream a MPEG2 TS PAL Stream with crtmpserver
Related
here is my question:
I use Android Apis "MediaMuxer" to combine h264 stream and aac stream to mp4 file,when I want to stop record,so I call this:mMediaMuxer.stop(),and the mp4 file can play well.
but sometimes,there is an unexpected happen,like kill power or power is gone suddendly,so there is no time to call "mMediaMuxer.stop()",finally this file can not play anyway....
Is anybody know now to fix this problem? I want to play this video event didn't call "mMediaMuxer.stop()" this method... or there is other Apis or sdk can combine h264+aac stream well?
I want to play flash media server stream via gstreamer. My video is published from camera to FMS with h264 encoding (720x480 Main,3.0).
My command for ubuntu is:
gst-launch-1.0 rtmpsrc
location="rtmp://192.168.1.153:1935/appname/mp4:cameraFeed44.mp4
live=1" ! decodebin name=decoder decoder. ! queue ! videoconvert !
queue ! xvimagesink
For resolution 720x480 it throws:
ERROR: from element /GstPipeline:pipeline0/GstRTMPSrc:rtmpsrc0: Internal data flow error.
Additional debug info:
gstbasesrc.c(2812): gst_base_src_loop (): /GstPipeline:pipeline0/ GstRTMPSrc:rtmpsrc0:
streaming task paused, reason error (-5)
ERROR: pipeline doesn't want to preroll.
However it works fine for low resolution, like: 320x240. But I need even more than FullHD.
Thanks,
Stan
Is it possible for a Chrome extension to listen for streaming audio from any of the browser's tabs? I would like to capture the streaming audio data and then analyse it.
Thanks
You could try 3 ways, neither one does provide 100% guarantee to meet your needs.
Before going into more detailed descriptions, I must note that Chrome extensions do not provide convenient tools for working on per connection level - sufficiently low level, required for stream capturing. This is by design. This is why the 1-st way is:
To look at other browsers, for example Firefox, which provides low-level APIs for connections. They are already known to be used by similar extensions. You may have a look at MediaStealer. If you do not have a specific requirement to build your system on Chrome, you should possibly move to Firefox.
You can develop a Chrome extension, which intercepts HTTP-requests by means of webRequest API, analyses their headers and extracts media urls (such as containing audio/mpeg MIME-type, for example, in HTTP-headers). Just for a quick example of code you make look at the following SO question - How to change response header in Chrome. Having the url you may force appropriate media download as a file. It will land in default downloads folder and may have unfriendly name. (I made such an extension, but I do not have requirements for further processing). If you need to further process such files, it can be a challenge to monitor them in the folder, and run additional analysis in a separate program.
You may have a look at NPAPI plugins in general, and their streaming APIs in particular. I can imagine that you create a plugin registered for, again, audio/mpeg MIME-type, and receives the data via NPP_NewStream, NPP_WriteReady and NPP_Write methods. The plugin can be wrapped into a Chrome extension. Though I made NPAPI plugins, I never used this API, and I'm not sure it will work as expected. Nethertheless, I'm mentioning this possibility here for completenees. This method requires some coding other than web-coding, meaning C/C++. NB. NPAPI plugins are deprecated and not supported in Chrome since September 2015.
Taking into account that you have some external (to the extension) "fingerprinting service" in mind, which sounds like an intelligent data processing, you may be interested in building all the system out of a browser. For example, you could, possibly, involve a HTTP-proxy, saving media from passing traffic.
If you're writing a Chrome extension, you can use the Chrome tabCapture API to record audio.
chrome.tabCapture.capture({audio: true}, function(stream) {
var recorder = new MediaRecorder(stream);
[...]
});
The rest is left as an exercise to the reader; MDN has more documentation on how to use MediaRecorder.
When this question was asked in 2013, neither chrome.tabCapture nor MediaRecorder existed.
Mac OSX solution using soundflower: http://rogueamoeba.com/freebies/soundflower/
After installing soundflower it should appear as a separate audio device in the sound preferences (apple > system preferences > sound). Divert the computer's audio to the 2ch option (stereo, 16ch is surround), then inside a DAW, such as 'audacity', set the audio input as soundflower. Now the sound should be channeled to your DAW ready for recording.
Note: having diverted the audio from the internal speakers to soundflower you will only be able to hear the audio if the 'soundflowerbed' app is actually open. You know it's open if there's a 8 legged blob in the top right task bar. Clicking this icon gives you the sound flower options.
My privoxy has the following log:
2013-08-28 18:25:27.953 00002f44 Request: api.audioaddict.com/v1/di/listener_sessions.jsonp?_method=POST&callback=_AudioAddict_WP_ListenerSession_create&listener_session%5Bid%5D=null&listener_session%5Bis_premium%5D=false&listener_session%5Bmember_id%5D=null&listener_session%5Bdevice_id%5D=6&listener_session%5Bchannel_id%5D=178&listener_session%5Bstream_set_key%5D=webplayer&_=1377699927926
2013-08-28 18:25:27.969 0000268c Request: api.audioaddict.com/v1/ping.jsonp?callback=_AudioAddict_WP_Ping__ping&_=1377699927928
2013-08-28 18:25:27.985 00002d48 Request: api.audioaddict.com/v1/di/track_history/channel/178.jsonp?callback=_AudioAddict_TrackHistory_Channel&_=1377699927942
2013-08-28 18:25:54.080 00003360 Request: pub7.di.fm/di_progressivepsy_aac?type=.flv
So I got the stream url and record it:
D:\Profiles\user\temp>wget pub7.di.fm/di_progressivepsy_aac?type=.flv
--18:26:32-- http://pub7.di.fm/di_progressivepsy_aac?type=.flv
=> `di_progressivepsy_aac#type=.flv'
Resolving pub7.di.fm... done.
Connecting to pub7.di.fm[67.221.255.50]:80... connected.
HTTP request sent, awaiting response... 200 OK
Length: unspecified [video/x-flv]
[ <=> ] 1,234,151 8.96K/s
I got the file that can be reproduced in any multimedia pleer.
I'm trying to get a video stream RTP/RTCP using HTML5, the stream was generated by gstreamer. I used examples of gstreamer, so I can pass through RTP ports:5000, and RTCP:5001, and can receive streams using gstreamer. But using HTML5 could not receive them. So I tried to read a bit about HTML5 and saw that it can receive theora/ogg, webm/vp8, mp4/avc, and protocols may be, HTTP, RTP, RTCP, UDP, and others, but I could not use RTP, RTCP or UDP, HTTP only managed to receive. But I had a very satisfactory result using the VLC plugin for Mozilla Firefox, using the UDP protocol. I wonder if anyone has any tips, I do not want to use source files as src="/tmp/test.avi" needs to be a video stream that can be udp, RTP, RTCP. Thank you!
If you don't need to stream at low fps, you can use GStreamer to transcode your stream in MJPEG and stream it in TCP, and then use VLC to get this TCP stream and stream it to HTTP. It works very well (0.5 sec of latency), but if you decrease the fps (1 fps) VLC introduces a latency of around 11 sec.
Here are some test commands that should work out of the box, using the GStreamer videotestsrc :
GStreamer :
gst-launch -v videotestsrc horizontal-speed=1 ! deinterlace ! videorate ! videoscale ! video/x-raw-yuv, framerate=15/1, width=256,
height=144 ! jpegenc quality=20 ! multipartmux
boundary="--videoboundary" ! tcpserversink host=localhost port=3000
VLC :
vlc -vvv -I rc tcp://localhost:3000 --sout
'#standard{access=http{mime=multipart/x-mixed-replace;boundary=--7b3cc56e5f51db803f790dad720ed50a},mux=mpjpeg,dst=localhost:8081}'
then open a browser to http://localhost:8081 (or create an HTML page with an img tag whose "src" attribute is http://localhost:8081)
When will it be posible to exchange any kind of data via P2P with WebRTC?
I'm thinking about making a P2P bittorrent software, not to stream audio/video.
Thanks
There are some JS libraries: PeerJS and BoneValue for P2P data exchanging using WebRTC.
If you want to make a BitTorrent-like JS app for sending files, be aware that there are some already - BtAppJS, ShareFest, ShareIt, RTC-P2P, and P2P-Share.
This will available in Chrome using the DataChannel: http://webrtc-demos.appspot.com/html/dc1.html
You can see an overview in this talk from Google I/O: http://www.youtube.com/watch?v=E8C8ouiXHHk&t=24m30s
I have not been able to establish a data connection in Chrome 21 or 22 (Canary), next step is to try compiling for myself.
P2P bittorrent software based on WebRTC already exists. It's called WebTorrent (source).
WebRTC that doesn't stream audio/video but arbitrary data is called RTCDataChannel: https://developer.mozilla.org/en-US/docs/Web/API/RTCDataChannel
But you are late - WebTorrent already exists.
WebTorrent client: https://github.com/webtorrent/webtorrent
WebTorrent tracker implementation: https://github.com/Novage/wt-tracker
Hybrid BitTorrent\WebTorrent tracker: https://github.com/webtorrent/bittorrent-tracker