sending raw audio
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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 ------------------------------------------------------------------------------ Come build with us! The BlackBerry® Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9-12, 2009. Register now! http://p.sf.net/sfu/devconf _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
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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:[hidden email]] Sent: 23 September 2009 19:10 To: [hidden email] 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 ------------------------------------------------------------------------ ------ Come build with us! The BlackBerry® Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9-12, 2009. Register now! http://p.sf.net/sfu/devconf _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel ------------------------------------------------------------------------------ Come build with us! The BlackBerry® Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9-12, 2009. Register now! http://p.sf.net/sfu/devconf _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
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I have a Freescale iMX27.
--Ron On Thu, Sep 24, 2009 at 6:54 AM, Andy Maginnis <[hidden email]> 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:[hidden email]] > Sent: 23 September 2009 19:10 > To: [hidden email] > 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 > > ------------------------------------------------------------------------ > ------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and > stay > ahead of the curve. Join us from November 9-12, 2009. Register > now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > ------------------------------------------------------------------------------ Come build with us! The BlackBerry® Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9-12, 2009. Register now! http://p.sf.net/sfu/devconf _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
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Ok, I'm a little further now. I had an endianness issue that I was
able to get around thanks to some other recent discussions on streaming audio in the last couple days. I'm now fighting some serious crackling I get when capturing from the mic and streaming the result across the network. On the capture side, I run this: gst-launch-0.10 -v alsasrc ! audio/x-raw-int,channels=1 ! audioconvert ! audio/x-raw-int,channels=1,depth=16,width=16,rate=24000,endianness=4321 ! rtpL16pay ! udpsink host=192.168.17.81 port=5435 And then the receiver: gst-launch-0.10 -v udpsrc port=5435 caps ="application/x-rtp, media=(string)audio, clock-rate=(int)32000, 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, endianness=1234" ! rtpL16depay ! audioconvert ! alsasink sync=false If I capture to a WAV file, and then stream that file across the network, I don't get any crackle. If I do the above capture across the network, I can barely make out the words I'm saying because of the crackle. Now, I'm using a 400Mhz Freescale iMX27 ARM CPU. When I do captures, I get warning messages like these: WARNING: from element /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: Can't record audio fast enough Additional debug info: gstbaseaudiosrc.c(805): gst_base_audio_src_create (): /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: dropped 480 samples I think I get more of these messages when I do a straight capture to the network, so it seems likely that the crackling has to do with the dropped samples. The lower the sample rate, the more messages I receive. I can't even capture at 8000 or 16000. I tried to fake my way around this issue by writing to a named pipe as follows: gst-launch-0.10 -v alsasrc ! audio/x-raw-int,channels=1 ! audioconvert ! audio/x-raw-int,channels=1,depth=16,width=16,rate=24000 ! audioconvert ! wavenc ! filesink location=/tmp/whee gst-launch -v --gst-debug=GST_CAT:3 filesrc location=/tmp/whee ! audio/x-raw-int,rate=24000,width=16,depth=16,channels=1,signed=true,endianness=1234 ! audioconvert ! rtpL16pay ! udpsink host=192.168.17.81 port=5435 But this has just as much crackling as sending straight across the network. Does anyone have suggestions on pipeline changes to get rid of the crackling? --Ron On Thu, Sep 24, 2009 at 8:26 AM, Ron Yorgason <[hidden email]> wrote: > I have a Freescale iMX27. > > --Ron > > On Thu, Sep 24, 2009 at 6:54 AM, Andy Maginnis <[hidden email]> 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:[hidden email]] >> Sent: 23 September 2009 19:10 >> To: [hidden email] >> 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 >> >> ------------------------------------------------------------------------ >> ------ >> Come build with us! The BlackBerry® Developer Conference in SF, CA >> is the only developer event you need to attend this year. Jumpstart your >> developing skills, take BlackBerry mobile applications to market and >> stay >> ahead of the curve. Join us from November 9-12, 2009. Register >> now! >> http://p.sf.net/sfu/devconf >> _______________________________________________ >> gstreamer-devel mailing list >> [hidden email] >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> ------------------------------------------------------------------------------ >> Come build with us! The BlackBerry® Developer Conference in SF, CA >> is the only developer event you need to attend this year. Jumpstart your >> developing skills, take BlackBerry mobile applications to market and stay >> ahead of the curve. Join us from November 9-12, 2009. Register now! >> http://p.sf.net/sfu/devconf >> _______________________________________________ >> gstreamer-devel mailing list >> [hidden email] >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > ------------------------------------------------------------------------------ Come build with us! The BlackBerry® Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9-12, 2009. Register now! http://p.sf.net/sfu/devconf _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
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