title: Send PCM data to VLC player using gstreamer
body: <p>I want to send raw audio data(PCM) to VLC player on RTP for playing the PCM using gstreamer.</p>
  
  <p>Here is the command to send the PCM</p>
  
  <blockquote>
    <p>gst-launch-1.0 -v filesrc location=/home/webos/pcm_data_dump ! audio/x-raw, rate=44100, channels=2, endianness=1234, format=S16LE, layout=interleaved, clock-rate=44100 ! audioconvert ! audioresample ! audio/x-raw, rate=44100, channels=2, format=S32LE, layout=interleaved ! audioconvert ! rtpL16pay pt=10 ! application/x-rtp, pt=10, encoding-name=L16, payload=10, clock-rate=44100, channels=2 !  udpsink host=192.168.0.2 port=5555</p>
  </blockquote>
  
  <p>Here is the VLC option to receive the PCM</p>
  
  <blockquote>
    <p>rtp://192.168.0.2:5555</p>
  </blockquote>
  
  <p>VLC player can get the PCM from gstreamer, but it cannot play.
  VLC shows the debug message like below.
  Lastly "core debug: Buffering 0%" message is shown repeatedly in VLC debug message.</p>
  
  <pre><code>core debug: output 'f32l' 44100 Hz Stereo frame=1 samples/8 bytes 
  core debug: looking for audio volume module matching "any": 2 candidates
  core debug: using audio volume module "float_mixer"
  core debug: input 's16l' 44100 Hz Stereo frame=1 samples/4 bytes
  core debug: looking for audio filter module matching "scaletempo": 14         candidates
  scaletempo debug: format: 44100 rate, 2 nch, 4 bps, fl32
  scaletempo debug: params: 30 stride, 0.200 overlap, 14 search
  scaletempo debug: 1.000 scale, 1323.000 stride_in, 1323 stride_out, 1059 
  standing, 264 overlap, 617 search, 2204 queue, fl32 mode
  core debug: using audio filter module "scaletempo"
  core debug: conversion: 's16l'-&gt;'f32l' 44100 Hz-&gt;44100 Hz Stereo-&gt;Stereo
  core debug: looking for audio converter module matching "any": 12 candidates
  audio_format debug: s16l-&gt;f32l, bits per sample: 16-&gt;32
  core debug: using audio converter module "audio_format"
  core debug: conversion pipeline complete
  core debug: conversion: 'f32l'-&gt;'f32l' 44100 Hz-&gt;44100 Hz Stereo-&gt;Stereo
  core debug: Buffering 0%
  core debug: conversion pipeline complete
  core debug: looking for audio resampler module matching "any": 3 candidates
  core debug: Buffering 0%
  core debug: Buffering 0%
  core debug: Buffering 0%
  core debug: Buffering 0%
  core debug: Buffering 0%
  .......
  </code></pre>
  
  <p>And, the log below is shown once the gstreamer command to send PCM starts.
  Normally, gstreamer is blocked with this message"New clock: GstSystemClock" when command starts.</p>
  
  <pre><code>Setting pipeline to PAUSED ...
  Pipeline is PREROLLING ...
  /GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:src: caps = audio/x-    raw, format=(string)S32LE, layout=(string)interleaved, rate=(int)44100,     channels=(int)2
  /GstPipeline:pipeline0/GstQueue:queue0.GstPad:sink: caps = audio/x-raw,   format=(string)S32LE, layout=(string)interleaved, rate=(int)44100, channels=  (int)2
  /GstPipeline:pipeline0/GstQueue:queue0.GstPad:sink: caps = audio/x-raw,   format=(string)S32LE, layout=(string)interleaved, rate=(int)44100, channels= (int)2
  /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps =   audio/x-raw, layout=(string)interleaved, rate=(int)44100, format=(string)S16BE,                                             channels=(int)2, channel-mask=(bitmask)0x0000000000000003
  /GstPipeline:pipeline0/GstQueue:queue1.GstPad:sink: caps = audio/x-raw, layout=(string)interleaved, rate=(int)44100, format=(string)S16BE, channels=(int)2, channel-mask=(bitmask)0x0000000000000003
  /GstPipeline:pipeline0/GstQueue:queue1.GstPad:sink: caps = audio/x-raw, layout=(string)interleaved, rate=(int)44100, format=(string)S16BE, channels=(int)2, channel-mask=(bitmask)0x0000000000000003
  /GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0.GstPad:src: caps = application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L16, encoding-params=(string)2, channels=(int)2, payload=(int)10, ssrc=(uint)2226113402, timestamp-offset=(uint)1744959080, seqnum-offset=(uint)62815
  /GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L16, encoding-params=(string)2, channels=(int)2, payload=(int)10, ssrc=(uint)2226113402, timestamp-offset=(uint)1744959080, seqnum-offset=(uint)62815
  /GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0.GstPad:sink: caps = audio/x-raw, layout=(string)interleaved, rate=(int)44100, format=(string)S16BE, channels=(int)2, channel-mask=(bitmask)0x0000000000000003
  /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps = audio/x-raw, format=(string)S32LE, layout=(string)interleaved, rate=(int)44100, channels=(int)2
  /GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0: timestamp = 1744959080
  /GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0: seqnum = 62815
  Pipeline is PREROLLED ...
  Setting pipeline to PLAYING ...
  New clock: GstSystemClock
  Got EOS from element "pipeline0".
  Execution ended after 0:00:00.622147167
  Setting pipeline to PAUSED ...
  Setting pipeline to READY ...
  Setting pipeline to NULL ...
  Freeing pipeline ...
  </code></pre>
  
  <p>gst-launch-0.1 has no problem, only 1.0 has problem.
  Is there any problem?</p>