Send PCM data to VLC player using gstreamer
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?
If I replace your filesrc with audiotestsrc , the example works for me. Still let me point out some room for improvement.
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
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