[gst-devel] sending raw audio

Ron Yorgason yorgasor at gmail.com
Thu Sep 24 14:26:16 CEST 2009


I have a Freescale iMX27.

--Ron

On Thu, Sep 24, 2009 at 6:54 AM, Andy Maginnis <jam at smru.co.uk> wrote:
> Ron,
> What ARM are you using? We have a cortex A8,
> inside a OMAP3530 on a Gumstix Overo Water
> a
>
> -----Original Message-----
> From: Ron Yorgason [mailto:yorgasor at gmail.com]
> Sent: 23 September 2009 19:10
> To: gstreamer-devel at lists.sourceforge.net
> Subject: [gst-devel] sending raw audio
>
> I'm working on an audio/video streaming application on the ARM
> platform.  We don't seem to have enough CPU horsepower on these boards
> to capture and encode to MP3, so right now we're trying to send raw
> audio.
>
> To capture & stream, I'm using this command:
>
> gst-launch-0.10 -v  alsasrc !
> audio/x-raw-int,rate=24000,width=16,depth=16,channels=1,signed=true !
> audioconvert ! rtpL16pay  ! udpsink host=192.168.17.81 port=5435
>
> On the receiving side, I have this:
> gst-launch-0.10 -v udpsrc port=5435 caps ="application/x-rtp,
> media=(string)audio, clock-rate=(int)24000, encoding-name=(string)L16,
> encoding-params=(string)1, channels=(int)1, channel-positions=(int)1,
> payload=(int)96, ssrc=(guint)1168011267, clock-base=(guint)309599748,
> seqnum-base=(guint)27324"  !   rtpL16depay !  audioconvert ! alsasink
> sync=false
>
> When I test this between my desktop & laptop, it works great.  When I
> go from the ARM board to my desktop, it works ok.  But when I go from
> the ARM board to another ARM board, or my laptop to the ARM board, I
> hear the speakers turn on, but no sound comes out.  The output shows
> this:
>
> Setting pipeline to PAUSED ...
> Pipeline is live and does not need PREROLL ...
> Setting pipeline to PLAYING ...
> New clock: GstSystemClock
> /GstPipeline:pipeline0/GstRtpL16Depay:rtpl16depay0.GstPad:src: caps =
> audio/x-raw-int, endianness=(int)4321, signed=(boolean)true,
> width=(int)16, depth=(int)16, rate=(int)24000, channels=(int)1,
> channel-positions=(GstAudioChannelPosition)<
> GST_AUDIO_CHANNEL_POSITION_NONE >
> /GstPipeline:pipeline0/GstRtpL16Depay:rtpl16depay0.GstPad:sink: caps =
> application/x-rtp, media=(string)audio, clock-rate=(int)24000,
> encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1,
> channel-positions=(int)1, payload=(int)96, ssrc=(guint)1168011267,
> clock-base=(guint)309599748, seqnum-base=(guint)27324
> /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps
> = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true,
> width=(int)32, depth=(int)32, rate=(int)24000, channels=(int)1,
> channel-positions=(GstAudioChannelPosition)<
> GST_AUDIO_CHANNEL_POSITION_NONE >
> /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps
> = audio/x-raw-int, endianness=(int)4321, signed=(boolean)true,
> width=(int)16, depth=(int)16, rate=(int)24000, channels=(int)1,
> channel-positions=(GstAudioChannelPosition)<
> GST_AUDIO_CHANNEL_POSITION_NONE >
> /GstPipeline:pipeline0/GstAlsaSink:alsasink0.GstPad:sink: caps =
> audio/x-raw-int, endianness=(int)1234, signed=(boolean)true,
> width=(int)32, depth=(int)32, rate=(int)24000, channels=(int)1,
> channel-positions=(GstAudioChannelPosition)<
> GST_AUDIO_CHANNEL_POSITION_NONE >
>
> I couldn't find a good definition of what the
> GST_AUDIO_CHANNEL_POSITION_NONE means, and I wasn't able to set it to
> GST_AUDIO_CHANNEL_POSITION_MONO from the command line (it looks like I
> need to use python or C APIs to do that), but from what I can tell,
> this comes from multichannel support, and I have just specified a
> single audio channel.  Is this what is preventing me from hearing the
> audio?  If I capture to a WAV file, and then play it back afterwards,
> it sounds fine.  So I'm not sure why streaming is failing so badly.
> The playback process also dies with a "Terminated" message within a
> minute or two.
>
> The ARM boards are running gstreamer-0.10.22.  I see that there's been
> a couple revisions since then, and if I have to upgrade to make it
> work, I will.  But I'd rather see if there's a way I can make this
> version work.
>
> --Ron
>
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