[gst-devel] alsasrc ! alawenc ! rtppcmapay ! udpsink - can receive, record then playback, but not playback live.

[Viraj Karandikar]

Hi,
I dont think there is any bug in gstreamer.
I have successfully run "audio capture - UDP send" on one board and "UDP
receive - audio playout" on other board.
But we didnt specify ssrc, clock-base, seqnum-base etc.. parameters for
udpsrc.
Try this once (your command with minimal udpsrc parameters)

gst-launch-0.10 -v udpsrc  port=1234 caps="application/x-rtp,
media=(string)audio, clock-rate=(int)8000, encoding-name=(string)PCMA"  !
rtppcmadepay  ! audio/x-alaw, rate=8000, channels=1  ! alawdec  ! alsasink

Regards,
Viraj
On Mon, Oct 5, 2009 at 7:28 PM, john oliver <johnwpoliver at hotmail.com>wrote:

> Hi,
>
> I managed to play back the stream live using VLC ( buffer time 100ms) . On
> the server I'm using
>
>  gst-launch-0.10 -v filesrc location=./audiotest1.pcma blocksize=800
> do-timestamp=true ! audio/x-alaw, rate=8000, channels=1  ! queue !
> rtppcmapay   ! udpsink host=x.x.x.x port=1234
>
>
> Perhaps I can't play it back live using gstreamer because of a bug ?
>
> John
> ------------------------------
> From: johnwpoliver at hotmail.com
> To: gstreamer-devel at lists.sourceforge.net
> Date: Mon, 5 Oct 2009 00:20:39 +0100
> Subject: [gst-devel] alsasrc ! alawenc ! rtppcmapay ! udpsink - can
> receive, record then playback, but not playback live.
>
> Hi all,
>
> I'm trying to stream a live source encoded as pcm ulaw over rtp over udp.
> I'm able to send it out on one side using gst-launch,  and on the client
> side I can save the stream to a file, then play it back. But I'd like to be
> able to play it back live on the client side, which I'm unable to do.
>
> My server
>
>  gst-launch-0.10 -v alsasrc   do-timestamp=true  !  audioconvert !
> audioresample ! alawenc ! rtppcmapay  ! udpsink host=127.0.0.1 port=1234
>
> /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: actual-buffer-time = 200000
> /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: actual-latency-time = 10000
> /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0.GstPad:src: caps =
> audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)32,
> depth=(int)32, rate=(int)8000, channels=(int)2
> Pipeline is live and does not need PREROLL ...
> Setting pipeline to PLAYING ...
> New clock: GstAudioSrcClock
> [snip]
> /GstPipeline:pipeline0/GstRtpPmcaPay:rtppmcapay0.GstPad:sink: caps =
> audio/x-alaw, rate=(int)8000, channels=(int)1
> /GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps =
> application/x-rtp, media=(string)audio, clock-rate=(int)8000,
> encoding-name=(string)PCMA, payload=(int)8, ssrc=(guint)2636653180,
> clock-base=(guint)4163613013, seqnum-base=(guint)41237
>
> On the client
> receive and record:
>
> gst-launch-0.10 -v udpsrc port=1234 caps="application/x-rtp,
> media=(string)audio, clock-rate=(int)8000, encoding-name=(string)PCMA,
> payload=(int)8, ssrc=(guint)3682605153, clock-base=(guint)600325920,
> seqnum-base=(guint)34884" !  rtppcmadepay  ! filesink
> location=./received.pcma
>
> then playback:
>  gst-launch-0.10 -v filesrc location=./received.pcma  !  audio/x-alaw,
> rate=8000, channels=1 ! alawdec !  alsasink  blocksize=32
>
> The above works ok.
>
> But when I'm trying to use the following to playback live ..
>
>  gst-launch-0.10 -v udpsrc  do-timestamp=true port=1234
> caps="application/x-rtp, media=(string)audio, clock-rate=(int)8000,
> encoding-name=(string)PCMA, payload=(int)8, ssrc=(guint)3682605153,
> clock-base=(guint)600325920, seqnum-base=(guint)34884"  ! rtppcmadepay  !
> audio/x-alaw, rate=8000, channels=1  ! alawdec  ! alsasink
>
>
> I don't hear anything ( apart from the odd glitch) . It tells me ..
>
> Setting pipeline to PAUSED ...
> Pipeline is live and does not need PREROLL ...
> Setting pipeline to PLAYING ...
> New clock: GstSystemClock
> /GstPipeline:pipeline0/GstRtpPcmaDepay:rtppcmadepay0.GstPad:src: caps =
> audio/x-alaw, channels=(int)1, rate=(int)8000
> /GstPipeline:pipeline0/GstRtpPcmaDepay:rtppcmadepay0.GstPad:sink: caps =
> application/x-rtp, media=(string)audio, clock-rate=(int)8000,
> encoding-name=(string)PCMA, payload=(int)8, ssrc=(guint)3682605153,
> clock-base=(guint)600325920, seqnum-base=(guint)34884
> /GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:src: caps =
> audio/x-alaw, channels=(int)1, rate=(int)8000
> /GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:sink: caps =
> audio/x-alaw, channels=(int)1, rate=(int)8000
> /GstPipeline:pipeline0/GstALawDec:alawdec0.GstPad:src: caps =
> audio/x-raw-int, width=(int)16, depth=(int)16, endianness=(int)1234,
> signed=(boolean)true, rate=(int)8000, channels=(int)1
> /GstPipeline:pipeline0/GstALawDec:alawdec0.GstPad:sink: caps =
> audio/x-alaw, channels=(int)1, rate=(int)8000
> /GstPipeline:pipeline0/GstAlsaSink:alsasink0.GstPad:sink: caps =
> audio/x-raw-int, width=(int)16, depth=(int)16, endianness=(int)1234,
> signed=(boolean)true, rate=(int)8000, channels=(int)1
>
> This is the same caps I see when I playback from the recorded file, almost,
> where the alawdec sink buffer size is 4096 ..
>
>
>
> /GstPipeline:pipeline0/GstALawDec:alawdec0.GstPad:sink: caps =
> audio/x-alaw, rate=(int)8000, channels=(int)1
> alawdec: buffer size: 4096
> /GstPipeline:pipeline0/GstAlsaSink:alsasink0.GstPad:sink: caps =
> audio/x-raw-int, width=(int)16, depth=(int)16, endianness=(int)1234,
> signed=(boolean)true, rate=(int)8000, channels=(int)1
>
> Can anybody spot any glaring mistakes or perhaps give me any pointers,
> please.
>
> John
>
>
>
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-- 
- Viraj
Reality is merely an illusion, albeit a very persistent one.
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