Send PCM data to VLC player using gstreamer

Question

I want to send raw audio data(PCM) to VLC player on RTP for playing the PCM using gstreamer.

Here is the command to send the PCM

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

Here is the VLC option to receive the PCM

rtp://192.168.0.2:5555

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.

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'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo
core debug: looking for audio converter module matching "any": 12 candidates
audio_format debug: s16l->f32l, bits per sample: 16->32
core debug: using audio converter module "audio_format"
core debug: conversion pipeline complete
core debug: conversion: 'f32l'->'f32l' 44100 Hz->44100 Hz Stereo->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%
.......

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.

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 ...

gst-launch-0.1 has no problem, only 1.0 has problem. Is there any problem?

ensonic ensonic · Accepted Answer · 2015-09-15T15:47:26

If I replace your filesrc with audiotestsrc, the example works for me. Still let me point out some room for improvement.

use audioparse instead of the first capsfilter
don't audioconvert twice.

Here is a simplified pipeline that works for me:

gst-launch-1.0 -v audiotestsrc ! audioresample ! audioconvert ! rtpL16pay pt=10 ! application/x-rtp, pt=10, encoding-name=L16, payload=10, clock-rate=44100, channels=2 ! udpsink host=localhost port=5555

Send PCM data to VLC player using gstreamer

1 Answers