[gst-devel] udp transfer/play wav file problem

[ensonic]

hi,

does this help?

client terminal: gst-launch-0.10 -v udpsrc ! audioconvert ! alsasink

Stefan

On 10:35:35 pm 23/05/2006 sam wang <samwzm at yahoo.com> wrote:
> Hi, there,
>
> I am a newbie about Gstreamer. in the long run, I want to transfer
> sound file in the real time cross network. at the beginning, I want
> to have a test that transfers a wav file from server to client and
> then paly it on the client. I tried TCP first but out of lucky. in
> fact, I really want to use UDP or RTP instead TCP. since RTP is under
> development now, I guess I should use UDP.
>
> now I really try to get two simple working commands to test UDP.
> first I follows the sample from the UDP API. I running the following
> commands in the same machine from two terminals(client terminal is
> used to receive data from server, server terminal is used to send data
> to client. my gstreamer is 0.10):
>
> client terminal: gst-launch -v udpsrc ! fakesink dump=1
> server terminal: gst-launch -v audiotestsrc ! udpsink
>
> there is no problem, but I want to play the sound on the client
> machine instead of dumping it. so I modify the client command to
> alsasink as following:
>
> client terminal: gst-launch-0.10 -v udpsrc ! alsasink
> server terminal: gst-launch-0.10 -v audiotestsrc ! udpsink
>
> **********************************************************************
> ******************
>
> then the server terminal shows as following(same as fakesink dump=1)
> and hung there:
>
> Setting pipeline to PAUSED ...
> /pipeline0/audiotestsrc0.src: caps = audio/x-raw-int,
> endianness=(int)4321, signed=(boolean)true, width=(int)16,
> depth=(int)16, rate=(int)44100, channels=(int)1Pipeline is PREROLLING
> ... /pipeline0/udpsink0.sink: caps = audio/x-raw-int,
> endianness=(int)4321, signed=(boolean)true, width=(int)16,
> depth=(int)16, rate=(int)44100, channels=(int)1 Pipeline is PREROLLED
> ... Setting pipeline to PLAYING ...
> New clock: GstSystemClock
>
>
> while the client terminal shows as following and NO SOUND at all and
> then quit:
>
> Setting pipeline to PAUSED ...
> Pipeline is live and does not need PREROLL ...
> Setting pipeline to PLAYING ...
> New clock: GstSystemClock
> ERROR: from element /pipeline0/alsasink0: No standard error message
> for domain gst-stream-error-quark and code 11.
> Additional debug info:
> gstbaseaudiosink.c(427): gst_base_audio_sink_preroll ():
> /pipeline0/alsasink0: sink not negotiated.
> Execution ended after 12884000 ns.
> Setting pipeline to PAUSED ...
> Setting pipeline to READY ...
> Setting pipeline to NULL ...
> FREEING pipeline ...
>
> **********************************************************************
> ***************
>
> I don't know what's wrong here and what I should do to play it on
> client. the alsasink is no problem because I can use it to play a
> simple wav file. in fact, I am quite sure the client recieve
> correctly, because there is no problem if I first try to save to a file
> on the client and then play the file. the commands show as following:
>
> client terminal:  gst-launch-0.10 -v udpsrc ! filesink
> location=test.wav server terminal:  gst-launch-0.10 -v filesrc
> location=/usr/share/sounds/phone.wav ! udpsink
>
> client terminal:  gst-launch-0.10 filesrc location=test.wav !
> wavparse ! alsasink
>
> by the way, is there any good source/sample about how to use UDP/RTP?
> I really need that.
>
> any hint will great appriciate!
> Sammy
>
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