One microphone, recording mono fails
Locked
One microphone, recording mono fails
 
|
Hello,
I am unable to record only one channel (mono). I have only one microphone attached to sound card. But if I run this gst-launch gconfaudiosrc ! capsfilter \ caps=audio/x-raw-int,rate=44100,channels=1,depth=16 ! \ wavenc ! filesink location=test.wav then I get this error message: Setting pipeline to PAUSED ... ERROR: Pipeline doesn't want to pause. ERROR: from element /GstPipeline:pipeline0/GstGConfAudioSrc:gconfaudiosrc0/GstBin:bin0/GstAlsaSrc:alsasrc0: Could not negotiate format Additional debug info: gstbasesrc.c(2584): gst_base_src_start (): /GstPipeline:pipeline0/GstGConfAudioSrc:gconfaudiosrc0/GstBin:bin0/GstAlsaSrc:alsasrc0: Check your filtered caps, if any Setting pipeline to NULL ... Freeing pipeline ... It can be solved if I use two channels for recording, so this gst-launch gconfaudiosrc ! capsfilter \ caps=audio/x-raw-int,rate=44100,channels=2,depth=16 ! \ wavenc ! filesink location=test.wav works fine. But I would like to record one channel because further data processing is little easier. Thank you Jan Martinek ------------------------------------------------------------------------------ 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 |
Re: One microphone, recording mono fails
|
|
Hi,
You can capture 2 channels and then use audioconvert to convert it to mono channel.
gst-launch gconfaudiosrc ! capsfilter \
caps=audio/x-raw-int,rate=44100,channels=2,depth=16 ! \ audioconvert ! audio/x-raw-int,rate=8000,depth=16,channels=1,width=16,signed=\(boolean\)TRUE,endianness=\(int\)1234 ! wavenc ! filesink location=test.wav regards,
Viraj
On Tue, Sep 29, 2009 at 4:53 PM, Jan Martinek <[hidden email]> wrote: Hello, -- - Viraj Reality is merely an illusion, albeit a very persistent one. ------------------------------------------------------------------------------ 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 |
Re: One microphone, recording mono fails
|
|
Hi,
this is interesting, thank you. But is it a recommended, clean solution or just workaround? I have tried mono recording on a different (older) computer and it works fine there. Relevant line from "lspci" is this: 00:11.5 Multimedia audio controller: VIA Technologies, Inc. VT8233/A/8235/8237 AC97 Audio Controller (rev 50) On the other hand, one channel recording fails on my computer with 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) Does it depend on hardware? Do capabilities differ? I would be surprised, because this seems to me as very basic functionality. Thank you, Jan Martinek On 09/29/2009 01:57 PM, Viraj Karandikar wrote: > Hi, > You can capture 2 channels and then use audioconvert to convert it to > mono channel. > gst-launch gconfaudiosrc ! capsfilter \ > caps=audio/x-raw-int,rate=44100,channels=2,depth=16 ! \ > audioconvert ! > audio/x-raw-int,rate=8000,depth=16,channels=1,width=16,signed=\(boolean\)TRUE,endianness=\(int\)1234 > ! wavenc ! filesink location=test.wav > regards, > Viraj > > On Tue, Sep 29, 2009 at 4:53 PM, Jan Martinek <[hidden email] > <mailto:[hidden email]>> wrote: > > Hello, > > I am unable to record only one channel (mono). I have only one > microphone attached to sound card. But if I run this > > gst-launch gconfaudiosrc ! capsfilter \ > caps=audio/x-raw-int,rate=44100,channels=1,depth=16 ! \ > wavenc ! filesink location=test.wav > > then I get this error message: > > Setting pipeline to PAUSED ... > ERROR: Pipeline doesn't want to pause. > ERROR: from element > /GstPipeline:pipeline0/GstGConfAudioSrc:gconfaudiosrc0/GstBin:bin0/GstAlsaSrc:alsasrc0: > Could not negotiate format > Additional debug info: > gstbasesrc.c(2584): gst_base_src_start (): > /GstPipeline:pipeline0/GstGConfAudioSrc:gconfaudiosrc0/GstBin:bin0/GstAlsaSrc:alsasrc0: > Check your filtered caps, if any > Setting pipeline to NULL ... > Freeing pipeline ... > > > It can be solved if I use two channels for recording, so this > > gst-launch gconfaudiosrc ! capsfilter \ > caps=audio/x-raw-int,rate=44100,channels=2,depth=16 ! \ > wavenc ! filesink location=test.wav > > works fine. But I would like to record one channel because further data > processing is little easier. > > Thank you > Jan Martinek > > ------------------------------------------------------------------------------ > 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] > <mailto:[hidden email]> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > -- > - Viraj > Reality is merely an illusion, albeit a very persistent one. > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------------ > 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 |
Re: One microphone, recording mono fails
|
|
Hi,
I dont see any issue in using audioconvert.
Did you see the log by setting debug level to 5?
When exactly "Could not negotiate format" error is coming? it might give some clue on whats happening.
-Viraj
On Tue, Sep 29, 2009 at 5:55 PM, Jan Martinek <[hidden email]> wrote: Hi, -- - Viraj Reality is merely an illusion, albeit a very persistent one. ------------------------------------------------------------------------------ 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 |
Re: One microphone, recording mono fails
|
|
On Tue, 2009-09-29 at 18:43 +0530, Viraj Karandikar wrote:
> When exactly "Could not negotiate format" error is coming? it might > give some clue on whats happening. not-negotiated means there was a format incompatibility somewhere. In this case that could be a capsfilter with a format/filter that upstream can't output (e.g. if the source doesn't support recording in mono), or some other element receiving a stream in a format it can't handle. wavenc is particularly picky - you should definitly have an audioconvert in front of wavenc, whatever you do. Something like this should work: gst-launch-0.10 alsasrc ! 'audio/x-raw-int,channels=1;audio/x-raw-int' ! audioconvert ! audio/x-raw-int,channels=1,depth=16 ! wavenc ! filesink location=foo.wav Cheers -Tim ------------------------------------------------------------------------------ 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 |
|
|
In reply to this post by Viraj Karandikar
I think this is just defaulting to your HW capabilities. Use –v
to see whats happening. examples on my machine, gst-launch -v alsasrc device=hw:1 ! audioconvert !
alsasink device=hw:1 gst-launch -v alsasrc device=hw:0 ! audioconvert !
alsasink device=hw:0 produce differing numbers of source channels. I suspect your
older computer only has one channel. >LAUNCH 1 /pipeline0/alsasrc0.src: caps = audio/x-raw-int,
endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16,
rate=(int)48000, channels=(int)1 ……. /pipeline0/audioconvert0.src: caps = audio/x-raw-int,
endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16,
rate=(int)48000, channels=(int)2 /pipeline0/audioconvert0.sink: caps = audio/x-raw-int,
endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16,
rate=(int)48000, channels=(int)1 /pipeline0/alsasink0.sink: caps = audio/x-raw-int,
endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16,
rate=(int)48000, channels=(int)2 >LAUNCH 2 pipeline0/alsasrc0.src: caps = audio/x-raw-int,
endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)32,
rate=(int)44100, channels=(int)2 …. /pipeline0/audioconvert0.src: caps = audio/x-raw-int,
endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)32,
rate=(int)44100, channels=(int)2 /pipeline0/audioconvert0.sink: caps = audio/x-raw-int,
endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)32,
rate=(int)44100, channels=(int)2 New clock: GstAudioSrcClock /pipeline0/alsasink0.sink: caps = audio/x-raw-int,
endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)32,
rate=(int)44100, channels=(int)2/ I can do gst-launch -v alsasrc device=hw:0 ! alsasink device=hw:0 but not gst-launch -v alsasrc device=hw:1 ! alsasink device=hw:1 as I need to modify the number of channels, I need the
audioconvert for this one! From: Viraj Karandikar
[mailto:[hidden email]] Hi, I dont see any issue in using audioconvert. Did you see the log by setting debug level to 5? When exactly "Could not negotiate format" error is
coming? it might give some clue on whats happening. -Viraj On Tue, Sep 29, 2009 at 5:55 PM, Jan Martinek <[hidden email]> wrote: Hi,
> <mailto:[hidden email]>>
wrote: > <mailto:[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 |
| Free forum by Nabble | Edit this page |