[Speex-dev] error during make while installing Linphone-1.5.1
venkatram mustoor
mustoor.venkatram at gmail.com
Thu Feb 15 20:38:48 PST 2007
Hi All,
I am getting this error during make.
please help me./
speexec.c: In function `speex_ec_process':
speexec.c:112: syntax error before "noise"
cc1: warnings being treated as errors
speexec.c:133: warning: implicit declaration of function
`speex_echo_state_reset'
speexec.c:148: warning: passing arg 5 of `speex_echo_cancel' makes
pointer from integer without a cast
speexec.c:149: warning: comparison between pointer and integer
speexec.c:150: warning: passing arg 3 of `speex_preprocess' makes
pointer from integer without a cast
make[3]: *** [speexec.lo] Error 1
make[3]: Leaving directory
`/home/umesh/IPPHONE/linphone-1.5.1/mediastreamer2/src'
make[2]: *** [all] Error 2
make[2]: Leaving directory
`/home/umesh/IPPHONE/linphone-1.5.1/mediastreamer2/src'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/home/umesh/IPPHONE/linphone-1.5.1/mediastreamer2'
make: *** [all] Error 2
This is the second time i putting this error, please help me.
On 2/6/07, speex-dev-request at xiph.org <speex-dev-request at xiph.org> wrote:
> Send Speex-dev mailing list submissions to
> speex-dev at xiph.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.xiph.org/mailman/listinfo/speex-dev
> or, via email, send a message with subject or body 'help' to
> speex-dev-request at xiph.org
>
> You can reach the person managing the list at
> speex-dev-owner at xiph.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of Speex-dev digest..."
>
>
> Today's Topics:
>
> 1. RE: Speex and RTP (David Hogan)
> 2. Re: Speex and RTP (Jean-Marc Valin)
> 3. Call for Xiph online meeting in 7th Feb (this Wednesday) at
> 18:00 UTC ( Ivo Emanuel Gon?alves )
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Mon, 5 Feb 2007 10:25:37 +1100
> From: "David Hogan" <david.hogan at freshtel.net>
> Subject: RE: [Speex-dev] Speex and RTP
> To: "Randy Schefiele" <scheifele2001 at yahoo.com>, <speex-dev at xiph.org>
> Message-ID:
> <971FA0386407DE4096E96382F4E3AD4443CA31 at Mail01.mel.au.office.freshtel.net>
>
> Content-Type: text/plain; charset="us-ascii"
>
> Hi Randy,
>
> One thing I would note is that speex is designed for 8kHz, 16kHz or
> 32kHz. 160 samples is equal to 20ms of 8kHz audio. Have you tried
> resampling from 11.025kHz to 8kHz and then using the speex 8kHz mode?
> (nb, or narrow band). Or, if you want to preserve the higher quality of
> your 11.025kHz sample rate, resample to 16kHz and use the wideband speex
> encoder.
>
> I believe there is new resampling functionality in the speex svn head,
> although I haven't tested it yet. You might also want to check out
> 'Secret Rabbit Code' for your resampling.
>
> Hope this helps.
> Dave
>
>
> ________________________________
>
> From: speex-dev-bounces at xiph.org
> [mailto:speex-dev-bounces at xiph.org] On Behalf Of Randy Schefiele
> Sent: Saturday, 3 February 2007 7:12 AM
> To: speex-dev at xiph.org
> Subject: [Speex-dev] Speex and RTP
>
>
> Hi -
>
> I am currently developing a RTSP/RTP/SDP solution to stream
> Speex encoded data. Using my current source, I have successfully
> streamed u-law and PCM encoded audio but have been unsuccessful thus far
> with Speex.
>
> Because of some constraints of my system, I am encoding audio at
> 11.025kHz. I am still using the 160 samples per frame which makes my
> frame size 28 bytes. I have successfully written the SDP needed to
> describe the Speex stream but I haven't found a player that will play
> the stream. I could be doing something wrong, but does anyone know of
> any players that can play RTP Speex streams so I can be sure that my
> solution is incorrect?
>
> Also, I have looked over the draft RFC of the Speex RTP payload
> and I was curious to know if, because I am not using a standard sample
> rate, my solution will work. In the draft RFC (and the Speex manual for
> that matter) it always refers to Speex frames containing 20ms of audio.
> In my case, a Speex frame actually contains roughly 14.5 seconds of
> audio. Am I going to need to hack the Speex encoder to make my frames
> be 20ms in order for the RTP to successfully play?
>
> Any advice/help anyone can give would be greatly appreciated.
>
> Thanks!
> Randy
>
>
>
> ________________________________
>
> The fish are biting.
> Get more visitors
> <http://us.rd.yahoo.com/evt=49679/*http://searchmarketing.yahoo.com/arp/
> sponsoredsearch_v2.php?o=US2140&cmp=Yahoo&ctv=Q107Tagline&s=Y&s2=EM&b=50
> > on your site using Yahoo! Search Marketing.
> <http://us.rd.yahoo.com/evt=49679/*http://searchmarketing.yahoo.com/arp/
> sponsoredsearch_v2.php?o=US2140&cmp=Yahoo&ctv=Q107Tagline&s=Y&s2=EM&b=50
> >
>
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20070205/bf8a6a16/attachment-0001.html
>
> ------------------------------
>
> Message: 2
> Date: Mon, 05 Feb 2007 21:26:30 +1100
> From: Jean-Marc Valin <jean-marc.valin at usherbrooke.ca>
> Subject: Re: [Speex-dev] Speex and RTP
> To: David Hogan <david.hogan at freshtel.net>
> Cc: speex-dev at xiph.org
> Message-ID: <45C70656.9090401 at usherbrooke.ca>
> Content-Type: text/plain; charset=ISO-8859-1
>
> > I believe there is new resampling functionality in the speex svn head,
> > although I haven't tested it yet. You might also want to check out
> > 'Secret Rabbit Code' for your resampling.
>
> Yes, I've just been working on a resampler recently. Its changing a lot,
> but it's now usable. I'd actually be quite happy to have some feedback
> on it.
>
> Jean-Marc
>
>
> > Hope this helps.
> > Dave
> >
> >
> > ________________________________
> >
> > From: speex-dev-bounces at xiph.org
> > [mailto:speex-dev-bounces at xiph.org] On Behalf Of Randy Schefiele
> > Sent: Saturday, 3 February 2007 7:12 AM
> > To: speex-dev at xiph.org
> > Subject: [Speex-dev] Speex and RTP
> >
> >
> > Hi -
> >
> > I am currently developing a RTSP/RTP/SDP solution to stream
> > Speex encoded data. Using my current source, I have successfully
> > streamed u-law and PCM encoded audio but have been unsuccessful thus far
> > with Speex.
> >
> > Because of some constraints of my system, I am encoding audio at
> > 11.025kHz. I am still using the 160 samples per frame which makes my
> > frame size 28 bytes. I have successfully written the SDP needed to
> > describe the Speex stream but I haven't found a player that will play
> > the stream. I could be doing something wrong, but does anyone know of
> > any players that can play RTP Speex streams so I can be sure that my
> > solution is incorrect?
> >
> > Also, I have looked over the draft RFC of the Speex RTP payload
> > and I was curious to know if, because I am not using a standard sample
> > rate, my solution will work. In the draft RFC (and the Speex manual for
> > that matter) it always refers to Speex frames containing 20ms of audio.
> > In my case, a Speex frame actually contains roughly 14.5 seconds of
> > audio. Am I going to need to hack the Speex encoder to make my frames
> > be 20ms in order for the RTP to successfully play?
> >
> > Any advice/help anyone can give would be greatly appreciated.
> >
> > Thanks!
> > Randy
> >
> >
> >
> > ________________________________
> >
> > The fish are biting.
> > Get more visitors
> > <http://us.rd.yahoo.com/evt=49679/*http://searchmarketing.yahoo.com/arp/
> > sponsoredsearch_v2.php?o=US2140&cmp=Yahoo&ctv=Q107Tagline&s=Y&s2=EM&b=50
> >> on your site using Yahoo! Search Marketing.
> > <http://us.rd.yahoo.com/evt=49679/*http://searchmarketing.yahoo.com/arp/
> > sponsoredsearch_v2.php?o=US2140&cmp=Yahoo&ctv=Q107Tagline&s=Y&s2=EM&b=50
> >
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > Speex-dev mailing list
> > Speex-dev at xiph.org
> > http://lists.xiph.org/mailman/listinfo/speex-dev
>
>
> ------------------------------
>
> Message: 3
> Date: Mon, 5 Feb 2007 17:40:32 +0000
> From: " Ivo Emanuel Gon?alves " <justivo at gmail.com>
> Subject: [Speex-dev] Call for Xiph online meeting in 7th Feb (this
> Wednesday) at 18:00 UTC
> To: Vorbis at xiph.org, vorbis-dev at xiph.org, flac-dev at xiph.org,
> Speex-dev at xiph.org, theora-dev at xiph.org, "New playlist format
> discussion" <playlist at lists.musicbrainz.org>, advocacy at xiph.org,
> ogg-dev at xiph.org
> Message-ID:
> <dc107ee70702050940t5d7e4f7tda7da6dfe93a40f2 at mail.gmail.com>
> Content-Type: text/plain; charset=UTF-8; format=flowed
>
> Agenda
>
> * aoTuV Release 1 merge
> * trac / trac cleaning
> * hosting for SpreadOgg project
> * Status of projects? libogg2
> * How can we make the services (esp. mailing lists and svn) more stable?
> * Will the Ambisonics discussion lead anywhere good?
> * contributors/webmasters
> * can someone bother to create a BSD-licensed libXSPF library to
> be hosted on Xiph?
>
> The agenda may be edited at http://wiki.xiph.org/index.php/MonthlyMeeting200702
>
> Is the time convenient for everyone?
>
> -Ivo
>
>
> ------------------------------
>
> _______________________________________________
> Speex-dev mailing list
> Speex-dev at xiph.org
> http://lists.xiph.org/mailman/listinfo/speex-dev
>
>
> End of Speex-dev Digest, Vol 33, Issue 5
> ****************************************
>
More information about the Speex-dev
mailing list