[Speex-dev] Re: Speex-dev Digest, Vol 33, Issue 9

venkatram mustoor mustoor.venkatram at gmail.com
Tue Feb 13 04:51:32 PST 2007

Hi All,

I am trying to cross compile speex-1.1.12 to powerpc-405,

i get a error after the make,   speexec.lo error,
please help me how to get rid of this error.

On 2/9/07, speex-dev-request at xiph.org <speex-dev-request at xiph.org> wrote:
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> Today's Topics:
>    1. Speex in C# Please help (Mohammed Ibrahim)
>    2. AEC and resample question (Jerry Trantow)
> ----------------------------------------------------------------------
> Message: 1
> Date: Thu, 8 Feb 2007 06:21:04 -0800 (PST)
> From: Mohammed Ibrahim <snouto1984 at yahoo.com>
> Subject: [Speex-dev] Speex in C# Please help
> To: speex-dev at xiph.org
> Message-ID: <935055.99042.qm at web58303.mail.re3.yahoo.com>
> Content-Type: text/plain; charset="iso-8859-1"
> hello guys in this forum.
>   i would like to use speex compression in my voice application.
>   but i have a problem .
>   i have an exception in C# once i try to use speex_lib_get_mode(int modein);
>   the exception message is "unable to find an entry point for the above function in the libspeex.dll"
>   my declaration for that function is:
>   [(DllImport(libspeex)]
>   public static extern IntPtr speex_lib_get_mode(int modein);
>   please if any one of you have a complete working source code of C# wrapper using speex please don't hesitate to mail it to me to check where is the error and why my compiler doesn't know an entry point for this file.
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> Message: 2
> Date: Thu, 8 Feb 2007 12:08:16 -0600
> From: "Jerry Trantow" <jtrantow at ieee.org>
> Subject: [Speex-dev] AEC and resample question
> To: <speex-dev at xiph.org>
> Message-ID: <004101c74bac$1ca17570$4001a8c0 at a64>
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> I understand that the capture/playback signals need to be sync'd for an AEC
> to adapt.  I'm a little bit confused on the requirements of synchronous
> sampling between the near end (mic/speaker) and the far end (phone line).  I
> have an embedded DSP system with mic and speaker getting 1msec packets
> containing 8 samples.  We can watch the DSP and ISDN clock frames drift and
> every few minutes we will drop or reuse a packet of samples.
> According to "Echo Cancellation Demystified" by Alexey Frunze
> http://www.spiritdsp.com/pdf/article_4.pdf (see section "Incorrect Codec
> Synchronization") dropping or reusing samples isn't a viable solution since
> it abruptly changes the echo path delay.  If I am dropping/reusing far end
> (phone line/ISDN) samples before they go out the speaker and after the mic,
> I don't understand why this has anything to do with the echo path.
> The solution in the paper is to put an adaptive SRC between the phone codec
> and mic/speaker codec.
> This must be a common situation.  I'm curious how people are implementing
> speex and the AEC?  Is there any benefit to resample?  Is the speex resample
> code suitable for adjusting the sample rates that only differ by clock
> crystal tolerances?
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> End of Speex-dev Digest, Vol 33, Issue 9
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