The VideoLAN Forums

alright, thank you

Hi all, I'm trying to stream some audio files with VLC 1.1.11 using the next command: :sout=#transcode{vcodec=none,acodec=mlaw,ab=192,channels=1,samplerate=24000}:rtp{dst=239.1.0.1,port=5004} :sout-keep which is an especial codification needed by some devices we use here. The audio sounds ok, howeve...

alright, thank you!

sorry, it's ugly_resampler... So, is this a known problem which has not been solved I guess...
Is there any way to avoid using this resampling module and letting ffmpeg to get control on all that stuff?

but VLC uses avcodec for resampling, doesn't it? (at least I've seen so at the messages) and so does ffmpeg... :S

I'm afraid that's not enough to get rid of this problem... The noise is still added when transcoding... I'm using the next sout string: :sout=#transcode{aenc=ffmpeg,acodec=ulaw,ab=64,channels=1,samplerate=8000}:rtp{dst=1.1.1.1, port=5004} :no-sout-rtp-sap :no-sout-standard-sap :ttl=1 :sout-keep And ...

I haven't tried yet, but shouldn't it be aenc? I'm just trying to transcode audio... and another little question, this shold be added at the sout string, right?

Thanks a lot Mr. Denis-Courmont.

Ben.

Hi all, I'm having some trouble with the transcoding. I need to stream ulaw audio via RTP protocol. I've tried to transcode the original mp3 files and it looks like it's working properly but when I listen to the stream sent, I can hear a periodic noise. To check wether the noise was coming at the tr...

this is what I'm sending: :sout=#rtp{dst=192.168.1.221,port=5004} :no-sout-rtp-sap :no-sout-standard-sap :ttl=1 :sout-keep just for trying I'm streaming from the same machine as the asterisk is running so network should not be an issue (I hope!) and this is the messages I get: main debug: looking fo...

but I'm streaming and getting the audio on the same machine...

I tried doing the same but without asterisk and in that case I get no error messages... so the problem must be somewhere between VLC and Asterisk... When writing to stdout or when reading from stdin...

It's quite strange...

Hi all, I'm using vlc 1.1.9 on an ubuntu 11.04 machine. I'm trying to get an rtp stream and send it to stdout where it will be used by asterisk for a music on hold application. Some time ago i got it working but now I started a new machine and I can't really remember what I did... :? The command I'm...

alright then,
thanks Rémi!

so what is it? how should I decode it?

Hi all, I got a small doubt about netsync... Is there any reason for using port 9875? When I capture the traffic with wireshark, it treats the netsync packets as SAPv1 or SAP/SDP packets but the info inside seems to be "malformed". How could I see what really is meant with this messages? T...

well, I guess that it wasn't a good idea to add the second part of the command at the more options. I've tried to write the directly in the prompt and now there are some UDP SAP/SDP packets in the wireshark. However, the synchronisation doesn't really work... :S it realizes that there's something go...

ok, I guess it's not working, since when I pause and resume the master the other clients don't synchronize... Isn't it as simple as adding the next lines to the stream clients? rtp://@:5004 and in the more options section --control netsync --netsync-master(-ip x.x.x.x ) should I add something else? ...

Hi all,
I'd like to know how can I make sure that netsync is working properly...
at the moment the streaming is playing synchronised but I'm trying to see UDP packets with wireshark and there's no sign of them...
how do the server and clients interact?

thank you!

ok, thx!

please, i need to get this working...
would it be enough to add something similar to http://git.videolan.org/?p=vlc.git;a=bl ... /rtp/rtp.c ?
or there must be much more code added?

Thank you!

Thanks Rémi, But if I choose to use that, it won't be compatible with my idea of 32 kHz samplerate. I'm having a look at the source code "modules/access/rtp/rtp.c" and it looks as if I could add a case for a known dynamic payload type (synchronising with "modules/services_discovery/sa...

Hello, I'm trying to use one of the pcm codecs (ulaw, alaw, s16b) to stream audio, but I'd like to change the samplerate to 32000 Hz. If I change this at the sout string,(transcode{acodec=s16b,ab=128,channels=2,samplerate=32000}) it uses a different RTP Payload Type value at SDP; 96 in this case. Ha...

Alright, I've seen that in the new versions of VLC it does support ulaw transcoding with RTP so... THANKS TO THE DEVELOPERS!

Thanks n_skid11,
but I've already had a look at this. That's why I'm asking if there's any posibility to get this work by adding some code or whatever...

sorry, for maybe becoming boring but any idea about how can I make this work?
It's necessary that I get this working....

Thank you!

Thank you Rémi, but RTP only supports PS/TS, doesn't it? I've tried PS and RAW and i get these error messages: main mux error: cannot add this stream stream_out_rtp stream out error: cannot add this stream to the muxer main packetizer error: cannot create packetizer output (mpga) Is it impossible to...