Doubts about gstreamer elements for playing and re send a radio stream

Ariel Argañaraz arielin82 at gmail.com
Mon Sep 16 14:28:10 PDT 2013


Hi all,

I was trying to play radio stream with gstreamer and I am a bit confused
with something.

The uri of the radio stream is
201.212.5.144/audio959-rockandpop?MSWMExt.asfand it is using MMS
protocol.

My first attempt was playing with this pipeline:  ( IT WORKS GREAT )

      GST_DEBUG=2 gst-launch uridecodebin uri=mmsh://
201.212.5.144/audio959-rockandpop?MSWMExt.asf ! decodebin2 ! autoaudiosink


And for record I use:
      GST_DEBUG=2 gst-launch uridecodebin uri=mmsh://
201.212.5.144/audio959-rockandpop?MSWMExt.asf ! decodebin2 ! filesink
location=/tmp/test.audio

This works too, but when it is recording the log shows that buffer is being
emptying and then they filled up again, it doesn't matter at the end
because the audio file it is successfully created.

The problem is that when I want to re-send the audio to my other
application That doesn't work.

So I decide to use the mmssrc element. And I tested again recording to a
file with this pipeline:
     GST_DEBUG=2 gst-launch mmssrc location=mmsh://
201.212.5.144/audio959-rockandpop?MSWMExt.asf ! decodebin2 ! filesink
location=/tmp/test.audio

And It works! Then I re-send to my other application and it works without
any problems.

I have realise that the mmssrc doesn't show in the logs that it is filling
and emptying the buffer as the uridecodebin shows.

* *And I would like to know if someone knows why the make different things
with the buffers or witch is better than the other?*


Here are the two pipelines with its logs that I told you:

with URIDECODEBIN:

GST_DEBUG=2 gst-launch uridecodebin uri=mmsh://
201.212.5.144/audio959-rockandpop?MSWMExt.asf ! decodebin2 ! filesink
location=/tmp/test.audio
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
0:00:00.094089755  3740       0xfb7630 WARN                asfdemux
gstasfdemux.c:2068:gst_asf_
demux_get_stream: Segment found for undefined stream: (1)
0:00:00.094172722  3740       0xfb7630 WARN                asfdemux
gstasfdemux.c:3090:gst_asf_demux_process_bitrate_props_object:<asfdemux0>
Stream id 1 wasn't found
0:00:02.383135605  3740 0x7f711c006850 WARN                  ffmpeg
gstffmpegcodecmap.c:140:gst_ff_channel_layout_to_gst: Unknown channels in
channel layout - assuming NONE layout
Prerolled, waiting for buffering to finish...
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
0:00:04.706824114  3740       0xfdf840 WARN                     bin
gstbin.c:2395:gst_bin_do_latency_func:<pipeline0> did not really configure
latency of 0:00:00.000000000
New clock: GstSystemClock

<<<<---------------HERE SHOWS THAT IT IS FILLING THE BUFFER!!!!

Buffering, setting pipeline to PAUSED ...
Prerolled, waiting for buffering to finish...
Done buffering, setting pipeline to PLAYING ...
0:00:08.134797738  3740       0xfdf840 WARN                     bin
gstbin.c:2395:gst_bin_do_latency_func:<pipeline0> did not really configure
latency of 0:00:00.000000000
Buffering, setting pipeline to PAUSED ...
Prerolled, waiting for buffering to finish...
Done buffering, setting pipeline to PLAYING ...
0:00:11.559758663  3740       0xfdf840 WARN                     bin
gstbin.c:2395:gst_bin_do_latency_func:<pipeline0> did not really configure
latency of 0:00:00.000000000
Buffering, setting pipeline to PAUSED ...
Prerolled, waiting for buffering to finish...


with MMSSRC:

ariel at omega:~ $ GST_DEBUG=2 gst-launch mmssrc location=mmsh://
201.212.5.144/audio959-rockandpop?MSWMExt.asf ! decodebin2 ! filesink
location=/tmp/test.audio
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
0:00:00.092874217  3660      0x1424770 WARN                asfdemux
gstasfdemux.c:2068:gst_asf_demux_get_stream: Segment found for undefined
stream: (1)
0:00:00.093138966  3660      0x1424770 WARN                asfdemux
gstasfdemux.c:3090:gst_asf_demux_process_bitrate_props_object:<asfdemux0>
Stream id 1 wasn't found
0:00:02.374889918  3660      0x1608590 WARN                  ffmpeg
gstffmpegcodecmap.c:140:gst_ff_channel_layout_to_gst: Unknown channels in
channel layout - assuming NONE layout
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
0:00:02.375598055  3660      0x160b270 WARN                     bin
gstbin.c:2395:gst_bin_do_latency_func:<pipeline0> did not really configure
latency of 0:00:00.000000000
New clock: GstSystemClock


sorry for my english,


-- 
Ariel Argañaraz
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